292 research outputs found
Autoencoding sensory substitution
Tens of millions of people live blind, and their number is ever increasing. Visual-to-auditory sensory substitution (SS) encompasses a family of cheap, generic solutions to assist the visually impaired by conveying visual information through sound. The required SS training is lengthy: months of effort is necessary to reach a practical level of adaptation. There are two reasons for the tedious training process: the elongated substituting audio signal, and the disregard for the compressive characteristics of the human hearing system.
To overcome these obstacles, we developed a novel class of SS methods, by training deep recurrent autoencoders for image-to-sound conversion. We successfully trained deep learning models on different datasets to execute visual-to-auditory stimulus conversion. By constraining the visual space, we demonstrated the viability of shortened substituting audio signals, while proposing mechanisms, such as the integration of computational hearing models, to optimally convey visual features in the substituting stimulus as perceptually discernible auditory components. We tested our approach in two separate cases. In the first experiment, the author went blindfolded for 5 days, while performing SS training on hand posture discrimination. The second experiment assessed the accuracy of reaching movements towards objects on a table. In both test cases, above-chance-level accuracy was attained after a few hours of training.
Our novel SS architecture broadens the horizon of rehabilitation methods engineered for the visually impaired. Further improvements on the proposed model shall yield hastened rehabilitation of the blind and a wider adaptation of SS devices as a consequence
Autoregressive Modelling of Hilbert Envelopes for Wide-band Audio Coding
Frequency Domain Linear Prediction (FDLP) represents the technique for approximating temporal envelopes of a signal using autoregressive models. In this paper, we propose a wide-band audio coding system exploiting FDLP. Specifically, FDLP is applied on critically sampled sub-bands to model the Hilbert envelopes. The residual of the linear prediction forms the Hilbert carrier, which is transmitted along with the envelope parameters. This process is reversed at the decoder to reconstruct the signal. In the objective and subjective quality evaluations, the FDLP based audio codec at kbps provides competitive results compared to the state-of-art codecs at similar bit-rates
Speech enhancement with frequency domain auto-regressive modeling
Speech applications in far-field real world settings often deal with signals
that are corrupted by reverberation. The task of dereverberation constitutes an
important step to improve the audible quality and to reduce the error rates in
applications like automatic speech recognition (ASR). We propose a unified
framework of speech dereverberation for improving the speech quality and the
ASR performance using the approach of envelope-carrier decomposition provided
by an autoregressive (AR) model. The AR model is applied in the frequency
domain of the sub-band speech signals to separate the envelope and carrier
parts. A novel neural architecture based on dual path long short term memory
(DPLSTM) model is proposed, which jointly enhances the sub-band envelope and
carrier components. The dereverberated envelope-carrier signals are modulated
and the sub-band signals are synthesized to reconstruct the audio signal back.
The DPLSTM model for dereverberation of envelope and carrier components also
allows the joint learning of the network weights for the down stream ASR task.
In the ASR tasks on the REVERB challenge dataset as well as on the VOiCES
dataset, we illustrate that the joint learning of speech dereverberation
network and the E2E ASR model yields significant performance improvements over
the baseline ASR system trained on log-mel spectrogram as well as other
benchmarks for dereverberation (average relative improvements of 10-24% over
the baseline system). The speech quality improvements, evaluated using
subjective listening tests, further highlight the improved quality of the
reconstructed audio.Comment: 10 page
Modeling and frequency tracking of marine mammal whistle calls
Submitted in partial fulfillment of the requirements for the degree of Master of Science at the Massachusetts Institute of Technology and the Woods Hole Oceanographic Institution February 2009Marine mammal whistle calls present an attractive medium for covert underwater
communications. High quality models of the whistle calls are needed in order to synthesize
natural-sounding whistles with embedded information. Since the whistle calls
are composed of frequency modulated harmonic tones, they are best modeled as a
weighted superposition of harmonically related sinusoids. Previous research with bottlenose
dolphin whistle calls has produced synthetic whistles that sound too “clean”
for use in a covert communications system. Due to the sensitivity of the human auditory
system, watermarking schemes that slightly modify the fundamental frequency
contour have good potential for producing natural-sounding whistles embedded with
retrievable watermarks. Structured total least squares is used with linear prediction
analysis to track the time-varying fundamental frequency and harmonic amplitude
contours throughout a whistle call. Simulation and experimental results demonstrate
the capability to accurately model bottlenose dolphin whistle calls and retrieve embedded
information from watermarked synthetic whistle calls. Different fundamental
frequency watermarking schemes are proposed based on their ability to produce natural
sounding synthetic whistles and yield suitable watermark detection and retrieval
Characterization and Decoding of Speech Representations From the Electrocorticogram
Millions of people worldwide suffer from various neuromuscular disorders such as amyotrophic lateral sclerosis (ALS), brainstem stroke, muscular dystrophy, cerebral palsy, and others, which adversely affect the neural control of muscles or the muscles themselves. The patients who are the most severely affected lose all voluntary muscle control and are completely locked-in, i.e., they are unable to communicate with the outside world in any manner. In the direction of developing neuro-rehabilitation techniques for these patients, several studies have used brain signals related to mental imagery and attention in order to control an external device, a technology known as a brain-computer interface (BCI). Some recent studies have also attempted to decode various aspects of spoken language, imagined language, or perceived speech directly from brain signals. In order to extend research in this direction, this dissertation aims to characterize and decode various speech representations popularly used in speech recognition systems directly from brain activity, specifically the electrocorticogram (ECoG). The speech representations studied in this dissertation range from simple features such as the speech power and the fundamental frequency (pitch), to complex representations such as the linear prediction coding and mel frequency cepstral coefficients. These decoded speech representations may eventually be used to enhance existing speech recognition systems or to reconstruct intended or imagined speech directly from brain activity. This research will ultimately pave the way for an ECoG-based neural speech prosthesis, which will offer a more natural communication channel for individuals who have lost the ability to speak normally
Deep Learning for Audio Signal Processing
Given the recent surge in developments of deep learning, this article
provides a review of the state-of-the-art deep learning techniques for audio
signal processing. Speech, music, and environmental sound processing are
considered side-by-side, in order to point out similarities and differences
between the domains, highlighting general methods, problems, key references,
and potential for cross-fertilization between areas. The dominant feature
representations (in particular, log-mel spectra and raw waveform) and deep
learning models are reviewed, including convolutional neural networks, variants
of the long short-term memory architecture, as well as more audio-specific
neural network models. Subsequently, prominent deep learning application areas
are covered, i.e. audio recognition (automatic speech recognition, music
information retrieval, environmental sound detection, localization and
tracking) and synthesis and transformation (source separation, audio
enhancement, generative models for speech, sound, and music synthesis).
Finally, key issues and future questions regarding deep learning applied to
audio signal processing are identified.Comment: 15 pages, 2 pdf figure
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