618 research outputs found
Wavenet based low rate speech coding
Traditional parametric coding of speech facilitates low rate but provides
poor reconstruction quality because of the inadequacy of the model used. We
describe how a WaveNet generative speech model can be used to generate high
quality speech from the bit stream of a standard parametric coder operating at
2.4 kb/s. We compare this parametric coder with a waveform coder based on the
same generative model and show that approximating the signal waveform incurs a
large rate penalty. Our experiments confirm the high performance of the WaveNet
based coder and show that the speech produced by the system is able to
additionally perform implicit bandwidth extension and does not significantly
impair recognition of the original speaker for the human listener, even when
that speaker has not been used during the training of the generative model.Comment: 5 pages, 2 figure
Using a low-bit rate speech enhancement variable post-filter as a speech recognition system pre-filter to improve robustness to GSM speech
Includes bibliographical references.Performance of speech recognition systems degrades when they are used to recognize speech that has been transmitted through GS1 (Global System for Mobile Communications) voice communication channels (GSM speech). This degradation is mainly due to GSM speech coding and GSM channel noise on speech signals transmitted through the network. This poor recognition of GSM channel speech limits the use of speech recognition applications over GSM networks. If speech recognition technology is to be used unlimitedly over GSM networks recognition accuracy of GSM channel speech has to be improved. Different channel normalization techniques have been developed in an attempt to improve recognition accuracy of voice channel modified speech in general (not specifically for GSM channel speech). These techniques can be classified into three broad categories, namely, model modification, signal pre-processing and feature processing techniques. In this work, as a contribution toward improving the robustness of speech recognition systems to GSM speech, the use of a low-bit speech enhancement post-filter as a speech recognition system pre-filter is proposed. This filter is to be used in recognition systems in combination with channel normalization techniques
Speech vocoding for laboratory phonology
Using phonological speech vocoding, we propose a platform for exploring
relations between phonology and speech processing, and in broader terms, for
exploring relations between the abstract and physical structures of a speech
signal. Our goal is to make a step towards bridging phonology and speech
processing and to contribute to the program of Laboratory Phonology. We show
three application examples for laboratory phonology: compositional phonological
speech modelling, a comparison of phonological systems and an experimental
phonological parametric text-to-speech (TTS) system. The featural
representations of the following three phonological systems are considered in
this work: (i) Government Phonology (GP), (ii) the Sound Pattern of English
(SPE), and (iii) the extended SPE (eSPE). Comparing GP- and eSPE-based vocoded
speech, we conclude that the latter achieves slightly better results than the
former. However, GP - the most compact phonological speech representation -
performs comparably to the systems with a higher number of phonological
features. The parametric TTS based on phonological speech representation, and
trained from an unlabelled audiobook in an unsupervised manner, achieves
intelligibility of 85% of the state-of-the-art parametric speech synthesis. We
envision that the presented approach paves the way for researchers in both
fields to form meaningful hypotheses that are explicitly testable using the
concepts developed and exemplified in this paper. On the one hand, laboratory
phonologists might test the applied concepts of their theoretical models, and
on the other hand, the speech processing community may utilize the concepts
developed for the theoretical phonological models for improvements of the
current state-of-the-art applications
Composition of Deep and Spiking Neural Networks for Very Low Bit Rate Speech Coding
Most current very low bit rate (VLBR) speech coding systems use hidden Markov
model (HMM) based speech recognition/synthesis techniques. This allows
transmission of information (such as phonemes) segment by segment that
decreases the bit rate. However, the encoder based on a phoneme speech
recognition may create bursts of segmental errors. Segmental errors are further
propagated to optional suprasegmental (such as syllable) information coding.
Together with the errors of voicing detection in pitch parametrization,
HMM-based speech coding creates speech discontinuities and unnatural speech
sound artefacts.
In this paper, we propose a novel VLBR speech coding framework based on
neural networks (NNs) for end-to-end speech analysis and synthesis without
HMMs. The speech coding framework relies on phonological (sub-phonetic)
representation of speech, and it is designed as a composition of deep and
spiking NNs: a bank of phonological analysers at the transmitter, and a
phonological synthesizer at the receiver, both realised as deep NNs, and a
spiking NN as an incremental and robust encoder of syllable boundaries for
coding of continuous fundamental frequency (F0). A combination of phonological
features defines much more sound patterns than phonetic features defined by
HMM-based speech coders, and the finer analysis/synthesis code contributes into
smoother encoded speech. Listeners significantly prefer the NN-based approach
due to fewer discontinuities and speech artefacts of the encoded speech. A
single forward pass is required during the speech encoding and decoding. The
proposed VLBR speech coding operates at a bit rate of approximately 360 bits/s
Syllable-based Pitch Encoding for Low Bit Rate Speech Coding with Recognition/Synthesis Architecture
Current HMM-based low bit rate speech coding systems work with phonetic vocoders. Pitch contour coding (on frame or phoneme level) is usually fairly orthogonal to other speech coding parameters. We make an assumption in our work that the speech signal contains supra-segmental cues. Hence, we present encoding of the pitch on the syllable level, used in the framework of a recognition/synthesis speech coder with phonetic vocoder. The results imply that high accuracy pitch contour reconstruction with negligible speech quality degradation is possible. The proposed pitch encoding technique operates on 30 - 35 bits per second
Recognizing Voice Over IP: A Robust Front-End for Speech Recognition on the World Wide Web
The Internet Protocol (IP) environment poses two relevant sources of distortion to the speech recognition problem: lossy speech coding and packet loss. In this paper, we propose a new front-end for speech recognition over IP networks. Specifically, we suggest extracting the recognition feature vectors directly from the encoded speech (i.e., the bit stream) instead of decoding it and subsequently extracting the feature vectors. This approach offers two significant benefits. First, the recognition system is only affected by the quantization distortion of the spectral envelope. Thus, we are avoiding the influence of other sources of distortion due to the encoding-decoding process. Second, when packet loss occurs, our front-end becomes more effective since it is not constrained to the error handling mechanism of the codec. We have considered the ITU G.723.1 standard codec, which is one of the most preponderant coding algorithms in voice over IP (VoIP) and compared the proposed front-end with the conventional approach in two automatic speech recognition (ASR) tasks, namely, speaker-independent isolated digit recognition and speaker-independent continuous speech recognition. In general, our approach outperforms the conventional procedure, for a variety of simulated packet loss rates. Furthermore, the improvement is higher as network conditions worsen.Publicad
Incremental Syllable-Context Phonetic Vocoding
Current very low bit rate speech coders are, due to complexity limitations, designed to work off-line. This paper investigates incremental speech coding that operates real-time and incrementally (i.e., encoded speech depends only on already-uttered speech without the need of future speech information). Since human speech communication is asynchronous (i.e., different information flows being simultaneously processed), we hypothesised that such an incremental speech coder should also operate asynchronously. To accomplish this task, we describe speech coding that reflects the human cortical temporal sampling that packages information into units of different temporal granularity, such as phonemes and syllables, in parallel. More specifically, a phonetic vocoder — cascaded speech recognition and synthesis systems — extended with syllable-based information transmission mechanisms is investigated. There are two main aspects evaluated in this work, the synchronous and asynchronous coding. Synchronous coding refers to the case when the phonetic vocoder and speech generation process depend on the syllable boundaries during encoding and decoding respectively. On the other hand, asynchronous coding refers to the case when the phonetic encoding and speech generation processes are done independently of the syllable boundaries. Our experiments confirmed that the asynchronous incremental speech coding performs better, in terms of intelligibility and overall speech quality, mainly due to better alignment of the segmental and prosodic information. The proposed vocoding operates at an uncompressed bit rate of 213 bits/sec and achieves an average communication delay of 243 ms
Progress report of a project in very low bit-rate speech coding
Background work in various levels of speech coding is reviewed, including unconstrained coding and recognition-synthesis approaches that assume the signal is speech. A pilot project in HMM-TTS based speech coding is then described, in which a comparison with harmonic plus noise modelling is also done. Results of the demonstration project including samples of speech under various transmission situations are presented in an accompanying web page. The report concludes by describing and enumerating the shortcomings of the demonstration system that define directions for future work. This work is a deliverable for the armasuisse funded project “RECOD - Low bit-rate speech coding
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