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Mikael Mieskolainen

dblp:145/5007 · DBLP profile ↗
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1ranked-venue papers
0as first author
0since 2021 · last 2013
—ORCID · none

Domains — the database's venue-derived domains; a paper can count in several

Artificial intelligence and machine learning · 1

Expertise — from the expertise taxonomy: the topics of the expert's papers under the CCF categories. A weight counts papers with recency: 1 for a paper about the topic, 0.3 when the topic is its context, halved every five years.

Computer graphics and multimedia
1 paper
Audio and music processing · 50% Multimedia systems and quality of experience · 50%

Topics — the 2 heaviest of 2, each with the papers that count most for it

TopicWeightPapersLastEvidence papers
Audio and music processing
microphone array processing
0.212013
Passive Temporal Offset Estimation of Multichannel Recordings of an Ad-Hoc Microphone Array · IEEE Trans. Speech Audio Process. 2013
Multimedia systems and quality of experience
multimedia synchronization
0.212013
Passive Temporal Offset Estimation of Multichannel Recordings of an Ad-Hoc Microphone Array · IEEE Trans. Speech Audio Process. 2013

Methods — techniques the papers use, named apart from their topics

nonlinear filtering · 0.2least squares estimation · 0.2cramer-rao lower bound · 0.2
YearPublicationVenuePosition
2013 Passive Temporal Offset Estimation of Multichannel Recordings of an Ad-Hoc Microphone Array
abstract
In recent years ad-hoc microphone arrays have become ubiquitous, and the capture hardware and quality is increasingly more sophisticated. Ad-hoc arrays hold a vast potential for audio applications, but they are inherently asynchronous, i.e., temporal offset exists in each channel, and furthermore the device locations are generally unknown. Therefore, the data is not directly suitable for traditional microphone array applications such as source localization and beamforming. This work presents a least squares method for temporal offset estimation of a static ad-hoc microphone array. The method utilizes the captured audio content without the need to emit calibration signals, provided that during the recording a sufficient amount of sound sources surround the array. The Cramer-Rao lower bound of the estimator is given and the effect of limited number of surrounding sources on the solution accuracy is investigated. A practical implementation is then presented using non-linear filtering with automatic parameter adjustment. Simulations over a range of reverberation and noise levels demonstrate the algorithm's robustness. Using smartphones an average RMS error of 3.5 samples (at 48 kHz) was reached when the algorithm's assumptions were met.
Pasi Pertilä, Matti S. Hämäläinen, Mikael Mieskolainen
IEEE Trans. Speech Audio Process.3