A view of Kanerva's sparse distributed memory

Pentti Kanerva is working on a new class of computers, which are called pattern computers. Pattern computers may close the gap between capabilities of biological organisms to recognize and act on patterns (visual, auditory, tactile, or olfactory) and capabilities of modern computers. Combinations of numeric, symbolic, and pattern computers may one day be capable of sustaining robots. The overview of the requirements for a pattern computer, a summary of Kanerva's Sparse Distributed Memory (SDM), and examples of tasks this computer can be expected to perform well are given

Parallel Auditory Filtering By Sustained and Transient Channels Separates Coarticulated Vowels and Consonants

A neural model of peripheral auditory processing is described and used to separate features of coarticulated vowels and consonants. After preprocessing of speech via a filterbank, the model splits into two parallel channels, a sustained channel and a transient channel. The sustained channel is sensitive to relatively stable parts of the speech waveform, notably synchronous properties of the vocalic portion of the stimulus it extends the dynamic range of eighth nerve filters using coincidence deteectors that combine operations of raising to a power, rectification, delay, multiplication, time averaging, and preemphasis. The transient channel is sensitive to critical features at the onsets and offsets of speech segments. It is built up from fast excitatory neurons that are modulated by slow inhibitory interneurons. These units are combined over high frequency and low frequency ranges using operations of rectification, normalization, multiplicative gating, and opponent processing. Detectors sensitive to frication and to onset or offset of stop consonants and vowels are described. Model properties are characterized by mathematical analysis and computer simulations. Neural analogs of model cells in the cochlear nucleus and inferior colliculus are noted, as are psychophysical data about perception of CV syllables that may be explained by the sustained transient channel hypothesis. The proposed sustained and transient processing seems to be an auditory analog of the sustained and transient processing that is known to occur in vision.Air Force Office of Scientific Research (F49620-92-J-0225); Advanced Research Projects Agency (AFOSR 90-0083, ONR N00014-92-J-4015); Office of Naval Research (N00014-95-I-0409

Silicon Neurons That Phase-Lock

We present a silicon neuron with a dynamic, active leak that enables precise spike-timing with respect to a time-varying input signal. Our neuron models the mammalian bushy cell, which enhances the phase-locking of its acoustically driven inputs. Our model enhances phase-locking by up to 38% (quantified by vector strength) across a 60 dB range of acoustic intensities, and up to 22% over a passive leak. Its conductance-based log-domain design yields a compact and efficient circuit, fabricated in 0.25 /spl mu/m CMOS, that is an ideal timing-enhancing component for neuromorphic speech recognition systems

Timescalenet: a multiresolution approcha for raw audio recognition

International audienceIn recent years, the use of Deep Learning techniques in audio signal processing has led the scientific community to develop machine learning strategies that allow to build efficient representations from raw waveforms for machine hearing tasks. In the present paper, we show the benefit of a multi-resolution approach : TimeScaleNet aims at learning an efficient representation of a sound, by learning time dependencies both at the sample level and at the frame level. At the sample level, TimeScaleNet's architecture introduces a new form of recurrent neural layer that acts as a learnable passband biquadratic digital IIR filterbank and self-adapts to the specific recognition task and dataset, with a large receptive field and very few learnable parameters. The obtained frame-level feature map is then processed using a residual network of depthwise separable atrous convolutions. This second scale of analysis allows to encode the time fluctuations at the frame timescale, in different learnt pooled frequency bands. In the present paper, TimeScaleNet is tested using the Speech Commands Dataset. We report a very high mean accuracy of 94.87±0.24% (macro averaged F1-score : 94.9 ± 0.24%) for this particular task

Stereophonic acoustic echo cancellation employing selective-tap adaptive algorithms

Published versio

Using hearing aid directional microphones and noise reduction algorithms to enhance cochlear implant performance

Abstract: Hearing aids and cochlear implants are two major hearing enhancement technologies but yet share little in research and development. The purpose of this study was to determine whether hearing aid directional microphones and noise reduction technologies could enhance cochlear implant users' speech understanding and ease of listening. Digital hearing aids serving as preprocessors were programmed to omni-directional microphone, directional microphone, and directional microphone plus noise reduction conditions. Three groups of subjects were tested with the hearing aid processed speech stimuli. Results indicated that hearing aids with directional microphones and noise reduction algorithms significantly enhanced speech understanding and listening comfort

A target guided subband filter for acoustic event detection in noisy environments using wavelet packets

This paper deals with acoustic event detection (AED), such as screams, gunshots, and explosions, in noisy environments. The main aim is to improve the detection performance under adverse conditions with a very low signal-to-noise ratio (SNR). A novel filtering method combined with an energy detector is presented. The wavelet packet transform (WPT) is first used for time-frequency representation of the acoustic signals. The proposed filter in the wavelet packet domain then uses a priori knowledge of the target event and an estimate of noise features to selectively suppress the background noise. It is in fact a content-aware band-pass filter which can automatically pass the frequency bands that are more significant in the target than in the noise. Theoretical analysis shows that the proposed filtering method is capable of enhancing the target content while suppressing the background noise for signals with a low SNR. A condition to increase the probability of correct detection is also obtained. Experiments have been carried out on a large dataset of acoustic events that are contaminated by different types of environmental noise and white noise with varying SNRs. Results show that the proposed method is more robust and better adapted to noise than ordinary energy detectors, and it can work even with an SNR as low as -15 dB. A practical system for real time processing and multi-target detection is also proposed in this work

Prosody takes over : towards a prosodically guided dialog system

The domain of the speech recognition and dialog system EVAR is train time table inquiry. We observed that in real human-human dialogs when the officer transmits the information, the customer very often interrupts. Many of these interruptions are just repetitions of the time of day given by the officer. The functional role of these interruptions is often determined by prosodic cues only. An important result of experiments where naive persons used the EVAR system is that it is hard to follow the train connection given via speech synthesis. In this case it is even more important than in human-human dialogs that the user has the opportunity to interact during the answer phase. Therefore we extended the dialog module to allow the user to repeat the time of day and we added a prosody module guiding the continuation of the dialog by analyzing the intonation contour of this utterance.Der Diskursbereich des Spracherkennungs- und Dialogsystems EVAR ist Fahrplanauskunft für Züge. Wir beobachteten, dass in realen Mensch-Mensch Dialogen der Kunde sehr oft den Auskunftsbeamten unterbricht, wenn dieser die Information übermittelt. Viele dieser Unterbrechungen sind ausschließlich Wiederholungen der Uhrzeitangabe des Beamten. Die funktionale Rolle dieser Unterbrechungen wird häufig alleine durch prosodische Mittel bestimmt. Ein wichtiges Ergebnis von Dialog Experimenten mit naiven Personen ergab, dass es schwer ist, den Verbindungsauskünften von EVAR via Sprachsynthese zu folgen. In diesem Fall ist es sogar noch wichtiger als in Mensch-Mensch Dialogen, dass der Benutzer die Möglichkeit hat, während der Antwortphase zu interagieren. Deshalb haben wir das Dialogmodul erweitert, um dem Benutzer die Möglichkeit zu geben, die Uhrzeitangaben zu wiederholen, und wir fügten ein Prosodiemodul hinzu, das die Fortführung des Dialogs steuert, indem die Intonation dieser Äußerung analysiert wir