Efficient pruning algorithm for hidden Markov model speech recognition

RETRACTED: Real time speech recognition algorithm on embedded system based on continuous Markov model

In a statistical based speech recognition system, one of the key issues is the selection of the Hidden Markov Model that best matches a given sequence of feature observations. The problem is usually addressed by the calculation of the maximum likelihood, ML, state sequence by means of a Viterbi or other decoder. Noise or inadequate training can produce a ML sequence associated with a Hidden Markov Model other than the correct model. The method of the present invention provides improved robustness by combining the standard ML state sequence score (416) with an additional path score (418) derived from the dynamics of the ML score as a function of time. These two scores, when combined, form a hybrid metric (420) that, when used with the decoder, optimizes selection of the correct Hidden Markov Model (422).

In this study, a real-time automatic speech recognition (ASR) system based on the Kaldi ASR toolkit, with low-latency and customizable models, without any internet connection, was developed. The proposed ASR system includes a voice activity detection (VAD) module and an audio transmitter as a front-end speech processing and a decoder for the received audio signals. The ASR system was evaluated in terms of ASR accuracy and speech processing speed. Consequently, the ASR system achieved high ASR accuracy on the CSJ (Corpus of Spontaneous Japanese) test set with super low-latency.

In addressing the problem of achieving high-accuracy real-time speech recognition systems, we focus on recognizing speech from ARPA's 20,000-word Wall Street Journal (WSJ) task, using current UNIX workstations. We have found that our standard approach---using a narrow beam width in a Viterbi search for simple discrete-density hidden Markov models (HMMs)---works in real time with only very low accuracy. Our most accurate algorithms recognize speech many times slower than real time. Our (yet unattained) goal is to recognize speech in real time at or near full accuracy.We describe the speed/accuracy trade-offs associated with several techniques used in a one-pass speech recognition framework:• Trade-offs associated with reducing the acoustic modeling resolution of the HMMs (e.g., output-distribution type, number of parameters, cross-word modeling)• Trade-offs associated with using lexicon trees, and techniques for implementing full and partial bigram grammars with those trees• Computation of Gaussian probabilities are the most time-consuming aspect of our highest accuracy system, and techniques allowing us to reduce the number of Gaussian probabilities computed with little or no impact on speech recognition accuracy.Our results show that tree-based modeling techniques used with appropriate acoustic modeling approaches achieve real-time performance on current UNIX workstations at about a 30% error rate for the WSJ task. The results also show that we can dramatically reduce the computational complexity of our more accurate but slower modeling alternatives so that they are near the speed necessary for real-time performance in a multipass search. Our near-future goal is to combine these two technologies so that real-time, high-accuracy large-vocabulary speech recognition can be achieved.

In this paper, we propose a novel boosted mixture learning (BML) framework for Gaussian mixture HMMs in speech recognition. BML is an incremental method to learn mixture models for classification problem. In each step of BML, one new mixture component is calculated according to functional gradient of an objective function to ensure that it is added along the direction to maximize the objective function the most. Several techniques have been proposed to extend BML from simple mixture models like Gaussian mixture model (GMM) to Gaussian mixture hidden Markov model (HMM), including Viterbi approximation to obtain state segmentation, weight decay to initialize sample weights to avoid overfitting, combining partial updating with global updating of parameters and using Bayesian information criterion (BIC) for parsimonious modeling. Experimental results on the WSJ0 task have shown that the proposed BML yields relative word and sentence error rate reduction of 10.9% and 12.9%, respectively, over the conventional training procedure.

In this paper, based on the study of two speech recognition algorithms, two designs of speech recognition system are given to realize this isolated speech recognition mobile robot control system based on ARM9 processor. The speech recognition process includes pretreatment of speech signal, characteristic extrication, pattern matching and post-processing. Mel-Frequency cepstrum coefficients (MFCC) and linear prediction cepstrum coefficients (LPCC) are the two most common parameters. Through analysis and comparison the parameters, MFCC shows more noise immunity than LPCC, so MFCC is selected as the characteristic parameters. Both dynamic time warping (DTW) and hidden markov model (HMM) are commonly used algorithm. For the different characteristics of DTW and HMM recognition algorithm, two different programs were designed for mobile robot control system. The effect and speed of the two speech recognition system were analyzed and compared.

It has further shaped the domain of real-time voice-controlled speech recognition systems. Things like language bias, computational expense and background noise have made strides more difficult in the past. This paper provides a novel view on these tasks, allowing for broader accessibility and real-world applicability of state-of-the-art models. We advocate a multi-dimensional methodology, including ad hoc model contextualization, tailored neural designs, and personalized learning strategies, to achieve voice and chip-height optimization. Existing speech recognition systems have a major limitation of being limited to few languages, dialects and accents. This study introduces a multilingual and multicultural model to address this issue and also provides access to the benefit of no-bias technologies to all parts of society. Out of these regional filters, it can work with more and work on them more precisely. In addition to this, the efficient structure also improves computational efficiency, meaning that the model is able to gain speed in real-time processing on low-power devices, fitting to rising demand for speech recognition in mobile and edge computing settings. Performance in contextual understanding remains a problem, errors in pronunciation or deviations from the dialect tend to lead to mistakes. As a result, the current study utilise semantic analysis and natural language processing (NLP) methods to assist understanding across different languages globally. These services are used in applications like medical/legal transcription or customer support, where correct transcription is critical. Additionally, the architecture enhances real-time processing by reducing latency and increasing responsiveness, which is critical in emergency response systems and autonomous vehicles where timely decision-making is crucial. By enhancing the efficiency and accuracy of ANN-based speech recognition, this research drives advancements in increasingly more accessible, effective, and reliable voice-activated technologies.

We present an end-of-utterance detector for real-time automatic speech recognition in far-field scenarios. The proposed system consists of three components: a long short-term memory (LSTM) neural network trained on acoustic features, an LSTM trained on l-best recognition hypotheses of the automatic speech recognition (ASR) decoder, and a feedforward deep neural network (DNN) combining embeddings derived from both LSTMs with pause duration features from the ASR decoder. At inference time, lower and upper latency (pause duration) bounds act as safeguards. Within the latency bounds, the utterance end-point is triggered as soon as the DNN posterior reaches a tuned threshold. Our experimental evaluation is carried out on real recordings of natural human interactions with voice-controlled far-field devices. We show that the acoustic embeddings are the single most powerful feature and particularly suitable for cross-lingual applications. We furthermore show the benefit of ASR decoder features, especially as a low cost alternative to ASR hypothesis em-beddings.

Speech recognition is one of the important areas in digital speech processing. The study of speech recognition is a part of a quest for artificially intelligent machines that can hear and understand spoken information. The conventional methods for speech recognition are very complicated and time consuming. To apply fuzzy logic to speech recognition is a new attempt in digital speech processing. The approach proposed in the paper simplifies the algorithm in speech recognition and makes the real-time processing time shorter. The situation considered in this paper is the simplest, i.e., the situation of speaker dependence, small vocabulary and isolated words.

This paper describes a new algorithm for speech recognition by using stereo vision pattern recognition equations with competition and cooperation. In our research, we applied recently developed 3-layered neural net (3LNN) equations to speech recognition. Our proposed acoustic models using these equations yield better recognition results than the hidden Markov model (HMM). When using a 216 (240) word database, stereo vision acoustic models gave 6.5% (6.6%) higher accuracy than HMMs.

We have developed a low-power VLSI chip for 60-kWord real-time continuous speech recognition based on a Hidden Markov Model (HMM). Our implementation includes a cache architecture using locality of speech recognition, beam pruning using a dynamic threshold, two-stage language model searching, highly parallel Gaussian Mixture Model (GMM) computation based on the mixture level, a variable 50-frame look-ahead scheme and elastic pipeline operation between the Viterbi transition and GMM processing. Results show that our implementation achieves 95% bandwidth reduction (70.86 MB/s) and 78% required frequency reduction (126.5 MHz) for 60-kWord real-time continuous speech recognition. The test chip, fabricated using 40 nm CMOS technology and containing 1.9 M transistors for logic and 7.8 Mbit on-chip memory, occupies 2.2 mm × 2.5 mm area. Measured data show 144 mW power consumption at 126.5 MHz and 1.1 V.

Algorithms for speech recognition can be characterized broadly as pattern recognition approaches and acoustic phonetic approaches. To date, the greatest degree of success in speech recognition has been obtained using pattern recognition paradigms. Thus, in this paper, we will be concerned primarily with showing how pattern recognition techniques have been applied to the problems of isolated word (or discrete utterance) recognition, connected word recognition, and continuous speech recognition. We will show that our understanding (and consequently the resulting recognizer performance) is best for the simplest recognition tasks and is considerably less well developed for large scale recognition systems.

Speech recognition in Human-Robot Interaction (HRI) fully relies on audio-based Automatic Speech Recognition. However, speech recognition that relies solely on audio faces significant challenges in noisy environments and may lead to poor performance in such environments. One approach to address this is to also use lipreading in combination with traditional speech recognition. Recent work has shown that audiovisual speech recognition (AVSR) can achieve a Word Error Rate (WER) of only 0.9% on the dataset LRS3. In this paper, we assess the potential of combining audio with lipreading on a social robot platform, Pepper, which has not yet been widely tested for AVSR. Given that prior research has focused on non-robotic domains, it remains unclear whether such models can generalize well to social robot environments. We systematically evaluate and compare the performance of established offline and real-time audiovisual models with their audio-only counterparts. The experiments were conducted in both a controlled laboratory setting and a dynamic and noisy public environment. We evaluated the data using WER and also measured the inference latency of real-time models via Real-Time Factor and Words Per Second rates. The results demonstrate real-time performance for audio-only speech recognition across all latency metrics and near real-time performance for models that combine audio with lipreading. We also explored factors that might influence the inference performance of these models to understand how much video contributes to the audio. This includes factors related to (1) environmental and temporal variations, (2) model behavior, and (3) implementation choices. Our findings indicate that for now the audio-only models outperform the audiovisual models on a social robot platform, in contrast to what has been reported in the benchmarked literature. We conclude that more work is still needed to benefit from lipreading in HRI.

Despite the maturity and availability of speech recognition systems, there are few available spiking speech recognition tasks that can be implemented with current neuromorphic systems. The methods used previously to generate spiking speech data are not capable of encoding speech in real-time or encoding very large modern speech datasets efficiently for input to neuromorphic processors. The ability to efficiently encode audio data to spikes will enable a wider variety of spiking audio datasets to be available and can also enable algorithmic development of real-time neuromorphic automatic speech recognition systems. Therefore, we developed speech2spikes, a simple and efficient audio processing pipeline that encodes recorded audio into spikes and is suitable for real-time operation with low-power neuromorphic processors. To demonstrate the efficacy of our method for audio to spike encoding we show that a small feed-forward spiking neural network trained on data generated with the pipeline achieves accuracy on the Google Speech Commands recognition task, exceeding the state-of-the art set by Spiking Speech Commands, a prior spiking encoding of the Google Speech Commands dataset, by over 10%. We also demonstrate a proof-of-concept real-time neuromorphic automatic speech recognition system using audio encoded with speech2spikes streamed to an Intel Loihi neuromorphic research processor.

The accurate recognition of speech is beneficial to the fields of machine translation and intelligent human–computer interaction. After briefly introducing speech recognition algorithms, this study proposed to recognize speech with a recurrent neural network (RNN) and adopted the connectionist temporal classification (CTC) algorithm to align input speech sequences and output text sequences forcibly. Simulation experiments compared the RNN-CTC algorithm with the Gaussian mixture model–hidden Markov model and convolutional neural network-CTC algorithms. The results demonstrated that the more training samples the speech recognition algorithm had, the higher the recognition accuracy of the trained algorithm was, but the training time consumption increased gradually; the more samples a trained speech recognition algorithm had to test, the lower the recognition accuracy and the longer the testing time. The proposed RNN-CTC speech recognition algorithm always had the highest accuracy and the lowest training and testing time among the three algorithms when the number of training and testing samples was the same.