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Audio signal processing

About: Audio signal processing is a research topic. Over the lifetime, 21463 publications have been published within this topic receiving 319597 citations. The topic is also known as: audio processing & Acoustic signal processing.


Papers
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Patent
02 Apr 2009
TL;DR: In this paper, an alternative scheme for minimizing feedback in audio processing systems is proposed, which is robust in situations with a high degree of correlation between the output signal and the input signal of an audio processing system.
Abstract: The invention relates to an audio processing system for processing an input sound to an output sound. The invention further relates to a method of estimating a feedback transfer function in an audio processing system. The object of the present invention is to provide an alternative scheme for minimizing feedback in audio processing systems. The problem is solved in that the audio processing system comprises an input transducer for converting an input sound to an electric input signal and defining an input side, an output transducer for converting a processed electric output signal to an output sound and defining an output side, a forward path being defined between the input transducer and the output transducer, and comprising a signal processing unit adapted for processing an SPU-input signal originating from the electric input signal and to provide a processed SPU-output signal, and an electric feedback loop from the output side to the input side comprising a feedback path estimation unit for estimating an acoustic feedback transfer function from the output transducer to the input transducer, and a enhancement unit for estimating noise-like signal components in the electric signal of the forward path and providing a noise signal estimate output, wherein the feedback path estimation unit is adapted to use the noise signal estimate output in the estimation of the acoustic feedback transfer function. This has the advantage of providing an adaptive feedback cancellation system which is robust in situations with a high degree of correlation between the output signal and the input signal of an audio processing system, e.g. a listening device. The invention may e.g. be used in public address systems, entertainment systems, hearing aids, head sets, mobile phones, wearable/portable communication devices, etc.

66 citations

Patent
28 Dec 2004
TL;DR: In this article, the authors propose a method for selecting an audio source when capturing audio data with at least one capture device, which is based on the first signal strengths of the audio signals detected through the audio input ports of the capture device.
Abstract: A method for selecting an audio source when capturing audio data with at least one capture device includes connecting an audio source to an audio input port of the capture device, turning off the output of the audio source, calculating first signal strengths of the audio signals detected through the audio input ports of the capture device, turning on the output of the audio source, calculating second signal strengths of the audio signals detected through the audio input ports of the capture device, calculating a signal change between the first signal strength and the second signal strength of each audio input port, and selecting the audio input port with the largest signal change.

66 citations

Journal ArticleDOI
TL;DR: A new architecture, design flow, and field-programmable gate array (FPGA) implementation analysis of a neuromorphic binaural auditory sensor, designed completely in the spike domain, is presented, allowing researchers to implement their own parameterized neuromorphic auditory systems in a low-cost FPGA in order to study the audio processing and learning activity that takes place in the brain.
Abstract: This paper presents a new architecture, design flow, and field-programmable gate array (FPGA) implementation analysis of a neuromorphic binaural auditory sensor, designed completely in the spike domain. Unlike digital cochleae that decompose audio signals using classical digital signal processing techniques, the model presented in this paper processes information directly encoded as spikes using pulse frequency modulation and provides a set of frequency-decomposed audio information using an address-event representation interface. In this case, a systematic approach to design led to a generic process for building, tuning, and implementing audio frequency decomposers with different features, facilitating synthesis with custom features. This allows researchers to implement their own parameterized neuromorphic auditory systems in a low-cost FPGA in order to study the audio processing and learning activity that takes place in the brain. In this paper, we present a 64-channel binaural neuromorphic auditory system implemented in a Virtex-5 FPGA using a commercial development board. The system was excited with a diverse set of audio signals in order to analyze its response and characterize its features. The neuromorphic auditory system response times and frequencies are reported. The experimental results of the proposed system implementation with 64-channel stereo are: a frequency range between 9.6 Hz and 14.6 kHz (adjustable), a maximum output event rate of 2.19 Mevents/s, a power consumption of 29.7 mW, the slices requirements of 11141, and a system clock frequency of 27 MHz.

66 citations

Proceedings ArticleDOI
12 May 2008
TL;DR: In this work, in order to detect the fundamental frequencies that are present in a signal, a set of candidates are selected from the spectrum, and all their possible combinations are generated and the best combination is chosen in a frame by frame analysis.
Abstract: The goal of a polyphonic music transcription system is to extract a score from an audio signal. A multiple fundamental frequency estimator is the main piece of these systems, whereas tempo detection and key estimation complement them to correctly extract the score. In this work, in order to detect the fundamental frequencies that are present in a signal, a set of candidates are selected from the spectrum, and all their possible combinations are generated. The best combination is chosen in a frame by frame analysis by applying a set of rules, taking into account the harmonic amplitudes and the spectral smoothness measure described in this work. The system was evaluated and compared to other works, yielding competitive results and performance.

66 citations

Journal ArticleDOI
TL;DR: A source separation evaluation revealed that when using large dictionaries, the proposed method produces a better separation quality in less time, and is up to 8 times faster than conventional algorithms.
Abstract: This paper proposes a computationally efficient algorithm for estimating the non-negative weights of linear combinations of the atoms of large-scale audio dictionaries, so that the generalized Kullback-Leibler divergence between an audio observation and the model is minimized. This linear model has been found useful in many audio signal processing tasks, but the existing algorithms are computationally slow when a large number of atoms is used. The proposed algorithm is based on iteratively updating a set of active atoms, with the weights updated using the Newton method and the step size estimated such that the weights remain non-negative. Algorithm convergence evaluations on representing audio spectra that are mixtures of two speakers show that with all the tested dictionary sizes the proposed method reaches a much lower value of the divergence than can be obtained by conventional algorithms, and is up to 8 times faster. A source separation evaluation revealed that when using large dictionaries, the proposed method produces a better separation quality in less time.

66 citations


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Performance
Metrics
No. of papers in the topic in previous years
YearPapers
202319
202263
2021217
2020525
2019659
2018597