A Bandpass Transform for Speaker Normalization

One of the major challenges for Automatic Speech Recognition is to handle speech variability. Inter-speaker variability is partly due to differences in speakers' anatomy and especially in their Vocal Tract geometry. Dissimilarities in Vocal Tract Length (VTL) are a known source of speech variation. Vocal Tract Length Normalization is a popular Speaker Normalization technique that can be implemented as a transformation of a spectrum frequency axis. We introduce in this document a new spectral transformation for Speaker Normalization. We use the Bilinear Transformation to introduce a new frequency warping resulting from a mapping of a prototype Band-Pass (BP) filter into a general BP filter. This new transformation called the Bandpass Transformation (BPT) offers two degrees of freedom enabling complex warpings of the frequency axis that are different from previous works with the Bilinear Transform. We then define a procedure to use BPT for Speaker Normalization based on the Nelder-Mead algorithm for the estimation of the BPT parameters. We present a detailed study of the performance of our new approach on two test sets with gender dependent and independent systems. Our results demonstrate clear improvements compared to standard methods used in VTL Normalization. A score compensation procedure is presented and results in further improvements of our results by refining our BPT parameter estimation

Towards an automatic speech recognition system for use by deaf students in lectures

According to the Royal National Institute for Deaf people there are nearly 7.5 million hearing-impaired people in Great Britain. Human-operated machine transcription systems, such as Palantype, achieve low word error rates in real-time. The disadvantage is that they are very expensive to use because of the difficulty in training operators, making them impractical for everyday use in higher education. Existing automatic speech recognition systems also achieve low word error rates, the disadvantages being that they work for read speech in a restricted domain. Moving a system to a new domain requires a large amount of relevant data, for training acoustic and language models. The adopted solution makes use of an existing continuous speech phoneme recognition system as a front-end to a word recognition sub-system. The subsystem generates a lattice of word hypotheses using dynamic programming with robust parameter estimation obtained using evolutionary programming. Sentence hypotheses are obtained by parsing the word lattice using a beam search and contributing knowledge consisting of anti-grammar rules, that check the syntactic incorrectness’ of word sequences, and word frequency information. On an unseen spontaneous lecture taken from the Lund Corpus and using a dictionary containing "2637 words, the system achieved 815% words correct with 15% simulated phoneme error, and 73.1% words correct with 25% simulated phoneme error. The system was also evaluated on 113 Wall Street Journal sentences. The achievements of the work are a domain independent method, using the anti- grammar, to reduce the word lattice search space whilst allowing normal spontaneous English to be spoken; a system designed to allow integration with new sources of knowledge, such as semantics or prosody, providing a test-bench for determining the impact of different knowledge upon word lattice parsing without the need for the underlying speech recognition hardware; the robustness of the word lattice generation using parameters that withstand changes in vocabulary and domain

Google USM: Scaling Automatic Speech Recognition Beyond 100 Languages

We introduce the Universal Speech Model (USM), a single large model that performs automatic speech recognition (ASR) across 100+ languages. This is achieved by pre-training the encoder of the model on a large unlabeled multilingual dataset of 12 million (M) hours spanning over 300 languages, and fine-tuning on a smaller labeled dataset. We use multilingual pre-training with random-projection quantization and speech-text modality matching to achieve state-of-the-art performance on downstream multilingual ASR and speech-to-text translation tasks. We also demonstrate that despite using a labeled training set 1/7-th the size of that used for the Whisper model, our model exhibits comparable or better performance on both in-domain and out-of-domain speech recognition tasks across many languages.Comment: 20 pages, 7 figures, 8 table

Quality of life in South Africa: the first ten years of democracy

South Africa celebrated ten years of democracy in 2004. This special issue of Social Indicators Research (SIR) reviews developments that have impacted on the quality of life of ordinary South Africans during the transition period. The issue updates an earlier volume of SIR (Volume 41) published in 1997 and as a stand-alone volume. The earlier volume was initiated following SIR editor Alex Michalos’ first visit to South Africa. This update on quality of life in South Africa follows on his return visit to the country in 2004 to see firsthand the changes that had occured in the meantime. This introductory article outlines major achievements of and setbacks for the new democracy and the challenges facing it in future. It provides the backround for the evaluations of a range of quality of life domains and issues including poverty and inequality, crime, the HIV/AIDS epidemic, migration and housing, religiosity, reconciliation, and optimism for the future which are covered in the eleven articles that follow. The introduction divides the articles under the headings of challenges, achievements, monitoring quality of life, and social capital for the future. The overview article concludes that improvements in quality of life have been uneven but goodwill and a positive outlook bode well for South African quality of life in future

A statistical approach to language modelling for the ATIS problem

Thesis (M. Eng.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 1995.Includes bibliographical references (p. 62-65).by Joshua D. Koppelman.M.Eng

A Vectorized Processing Algorithm for Continuous Speech Recognition and Associated FPGA-Based Architecture

This work analyzes Continuous Automatic Speech Recognition (CSR) and in contrast to prior work, it shows that the CSR algorithms can be specified in a highly parallel form. Through use of the MATLAB software package, the parallelism is exploited to create a compact, vectorized algorithm that is able to execute the CSR task. After an in-depth analysis of the SPHINX 3 Large Vocabulary Continuous Speech Recognition (LVCSR) engine the major functional units were redesigned in the MATLAB environment, taking special effort to flatten the algorithms and restructure the data to allow for matrix-based computations. Performing this conversion resulted in reducing the original 14,000 lines of C++ code into less then 200 lines of highly-vectorized operations, substantially increasing the potential Instruction Line Parallelism of the system. Using this vector model as a baseline, a custom hardware system was then created that is capable of performing the speech recognition task in real-time on a Xilinx Virtex-4 FPGA device. Through the creation independent hardware engines for each stage of the speech recognition process, the throughput of each is maximized by customizing the logic to the specific task. Further, a unique architecture was designed that allows for the creation of a static data path throughout the hardware, effectively removing the need for complex bus arbitration in the system. By making using of shared memory resources and applying a token passing scheme to the system, both the data movement within the design as well as the amount of active data are continually minimized during run-time. These results provide a novel method for perform speech recognition in both hardware and software, helping to further the development of systems capable of recognizing human speech