Equalization (audio)

Résumé

Equalization, or simply EQ, in sound recording and reproduction is the process of adjusting the volume of different frequency bands within an audio signal. The circuit or equipment used to achieve this is called an equalizer. Most hi-fi equipment uses relatively simple filters to make bass and treble adjustments. Graphic and parametric equalizers have much more flexibility in tailoring the frequency content of an audio signal. Broadcast and recording studios use sophisticated equalizers capable of much more detailed adjustments, such as eliminating unwanted sounds or making certain instruments or voices more prominent. Since equalizers "adjust the amplitude of audio signals at particular frequencies" they are, "in other words, frequency-specific volume knobs." Equalizers are used in recording and radio studios, production control rooms, and live sound reinforcement and in instrument amplifiers, such as guitar amplifiers, to correct or adjust the response of microphones, instrume

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https://en.wikipedia.org/wiki/Equalization_(audio)

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The acoustics of a room often refer to the quality of the room in terms of sound recording as well as sound reproduction. At low frequencies, where the wavelengths are of the same order of magnitude with the dimensions of the room, due to the standing waves phenomena, room modes are formed which can give rise to multiple irregularities and eventually damage the listening experience. In order to effectively address the problem at these frequencies, a thorough knowledge of the sound field in the room is often required. Through numerical studies, the sound field in the room could be simulated based on an FEM room model which would result in certain differences when comparing to the actual situation. On the other hand, the traditional experimental sampling of the sound field in a room often results in an impractical number of microphones.This thesis focuses on the development of a sound field reconstruction framework that makes use of certain sparse representation and recovery algorithm to reconstruct the sound field of the room at low frequencies using a limited set of microphones. The reconstruction framework is then tested under different conditions and scenarios to prove its validity. Once the performance of the framework has been validated through both simulation and actual measurements, the reconstruction framework is investigated in detail regarding its application in the field of active modal equalization. These active absorption methods are of high interest in the field of modal equalization due to the fact that passive absorption method at low frequencies are more often than not ineffective. The reconstruction framework is then used to first assess the spatial performance of the active electroacoustic absorbers which consists of a current-driven system of loudspeakers in enclosures that aims to dampen room modes by achieving optimal target impedance in front of the loudspeaker membranes. In addition, a more complex applications of the reconstruction framework in which its reconstructed sound field can be used to fine tune the control parameters of the active absorbers is also investigated. The research shows promising results in both application of the reconstruction frameworks and open doors to further development in both the field of sound field reconstruction as well as room modes equalization.

EPFL2021

Chargement

Hybrid NRZ/Multi-Tone Signaling for High-Speed Low-Power Wireline Transceivers

Kiarash Gharibdoust

Over the past few decades, incessant growth of Internet networking traffic and High-Performance Computing (HPC) has led to a tremendous demand for data bandwidth. Digital communication technologies combined with advanced integrated circuit scaling trends have enabled the semiconductor and microelectronic industry to dramatically scale the bandwidth of high-loss interfaces such as Ethernet, backplane, and Digital Subscriber Line (DSL). The key to achieving higher bandwidth is to employ equalization technique to compensate the channel impairments such as Inter-Symbol Interference (ISI), crosstalk, and environmental noise. Therefore, todayâs advanced input/outputs (I/Os) has been equipped with sophisticated equalization techniques to push beyond the uncompensated bandwidth of the system. To this end, process scaling has continually increased the data processing capability and improved the I/O performance over the last 15 years. However, since the channel bandwidth has not scaled with the same pace, the required signal processing and equalization circuitry becomes more and more complicated. Thereby, the energy efficiency improvements are largely offset by the energy needed to compensate channel impairments. In this design paradigm, re-thinking about the design strategies in order to not only satisfy the bandwidth performance, but also to improve power-performance becomes an important necessity. It is well known in communication theory that coding and signaling schemes have the potential to provide superior performance over band-limited channels. However, the choice of the optimum data communication algorithm should be considered by accounting for the circuit level power-performance trade-offs. In this thesis we have investigated the application of new algorithm and signaling schemes in wireline communications, especially for communication between microprocessors, memories, and peripherals. A new hybrid NRZ/Multi-Tone (NRZ/MT) signaling method has been developed during the course of this research. The system-level and circuit-level analysis, design, and implementation of the proposed signaling method has been performed in the frame of this work, and the silicon measurement results have proved the efficiency and the robustness of the proposed signaling methodology for wireline interfaces. In the first part of this work, a 7.5 Gb/s hybrid NRZ/MT transceiver (TRX) for multi-drop bus (MDB) memory interfaces is designed and fabricated in 40 nm CMOS technology. Reducing the complexity of the equalization circuitry on the receiver (RX) side, the proposed architecture achieves 1 pJ/bit link efficiency for a MDB channel bearing 45 dB loss at 2.5 GHz. The measurement results of the first prototype confirm that NRZ/MT serial data TRX can offer an energy-efficient solution for MDB memory interfaces. Motivated by the satisfying results of the first prototype, in the second phase of this research we have exploited the properties of multi-tone signaling, especially orthogonality among different sub-bands, to reduce the effect of crosstalk in high-dense wireline interconnects. A four-channel transceiver has been implemented in a standard CMOS 40 nm technology in order to demonstrate the performance of NRZ/MT signaling in presence of high channel loss and strong crosstalk noise. The proposed system achieves 1 pJ/bit power efficiency, while communicating over a MDB memory channel at 36 Gb/s aggregate data rate.

EPFL2016

Chargement

Spatial Acoustic Signal Processing

Mihailo Kolundzija

A sound field on a line or in a plane has an effectively limited spatial bandwidth determined by the temporal frequency. Similar can be said for sound fields from far-field sources when analyzed on circular and spherical apertures. Namely, for a given frequency and aperture size, a sound field is effectively composed of a finite number of circular or spherical harmonic components. Based on these two observations, it follows that if adequately sampled, sound fields can be represented and manipulated in a digital domain with negligible loss of information. The optimal sampling surface depends on the problem geometry, and the set of sampling points needs to be in accordance with the Nyquist criterion relative to the mentioned effective sound field bandwidth. In this thesis, we address the problems of sound field capture and reproduction from a practical perspective. More specifically, we present approaches that do not depend on acoustical models, but rely instead on obtaining an acoustic MIMO channel between transducers (microphones or loudspeakers) and a set of sampling (or control) points. Subsequently, sound field capture and reproduction are formulated as constrained optimization problems in a spatially discrete domain and solved using conventional numerical optimization tools. The first part of the thesis deals with spatial sound capture. We present a framework for analyzing and designing differential microphone arrays based on spatiotemporal sound field gradients. We also show how to record two- and three-dimensional sound fields with differential, circular, and spherical microphone arrays. Finally, we use the mentioned discrete optimization for computing filters for directional and sound field microphone arrays. In the second part of the thesis, we focus on spatial sound reproduction. We first present a design of a baffled loudspeaker array for reproducing sound with high directivity over a wide frequency range, which combines beamforming at low, and scattering from a rigid baffle at high frequencies. We next present Sound Field Reconstruction (SFR), which is an approach for optimally reproducing a desired sound field in a wide listening area by inverting a discrete, MIMO acoustic channel. In the end, we propose a single- and multi-channel low-frequency room equalization method, formulated as a discrete constrained optimization problem, with constraints designed to prevent excessive equalization filter gains, localization bias, and temporal distortions in the form of pre- and post-echos.

EPFL2012

Chargement

Spatial Acoustic Signal Processing

Mihailo Kolundzija

EPFL2012

Hybrid NRZ/Multi-Tone Signaling for High-Speed Low-Power Wireline Transceivers

Kiarash Gharibdoust

EPFL2016

EPFL2021