Zobrazeno 1 - 8
of 8
pro vyhledávání: '"Devangi N. Parikh"'
Publikováno v:
2022 IEEE International Parallel and Distributed Processing Symposium Workshops (IPDPSW).
Publikováno v:
Parallel Processing and Applied Mathematics ISBN: 9783030432287
PPAM (1)
PPAM (1)
Small prime-sized discrete Fourier transforms appear in various applications from quantum mechanics, material sciences and machine learning. The typical implementation of the discrete Fourier transform for such problem sizes is done as a cyclic convo
Externí odkaz:
https://explore.openaire.eu/search/publication?articleId=doi_________::71c4ffd21a11f188d43daa5a00ae6a1b
https://doi.org/10.1007/978-3-030-43229-4_15
https://doi.org/10.1007/978-3-030-43229-4_15
Publikováno v:
IEEE Transactions on Audio, Speech, and Language Processing. 20:1372-1382
Different noise-reduction methods have been proposed in the literature for single and multiple-microphone applications. For binaural hearing aids, multiple-microphone noise-reduction methods offer two significant psycho-acoustical advantages with res
Publikováno v:
INTERSPEECH
Autor:
David V. Anderson, Devangi N. Parikh
Publikováno v:
2011 Digital Signal Processing and Signal Processing Education Meeting (DSP/SPE).
In an environment with multiple audio sources, blind source separation (BSS) makes use of multiple microphone signals to estimate the respective source signals. Under normal circumstances, it is not possible to completely “unmix” the audio source
Autor:
David V. Anderson, Devangi N. Parikh
Publikováno v:
2010 Conference Record of the Forty Fourth Asilomar Conference on Signals, Systems and Computers.
Most of the state of the art speech noise suppression gains rely on the frequency decomposition of the signal. This decomposition is usually done using the fast Fourier transform. These algorithms may be successful in suppressing the noise but often
Publikováno v:
ICASSP
For cell-phone applications, single microphone noise suppression techniques have limited performance at very low SNR (close to 0 dB). In certain cases, they also suffer from the artifacts of nonlinear processing. In this paper, we will show that tech
Publikováno v:
WASPAA
In this paper we describe a technique that uses adaptive gain control to achieve noise suppression in speech signals. The method used to map the dynamic range of the signal is based on the human auditory perceptual model. Since the processing is base