Browsing by Author "Välimäki, Vesa, Prof., Aalto University, Finland"

Now showing 1 - 4 of 4
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    Analysis and Synthesis of Directional Reverberation
    (Aalto University, 2021) Alary, Benoit; Politis, Archontis, Dr., Tampere University, Finland; Schlecht, Sebastian J., Prof., Aalto University, Finland; Välimäki, Vesa, Prof., Aalto University, Finland; Signaalinkäsittelyn ja akustiikan laitos; Department of Signal Processing and Acoustics; Audio Signal Processing; Sähkötekniikan korkeakoulu; School of Electrical Engineering; Välimäki, Vesa, Prof., Aalto University, Department of Signal Processing and Acoustics, Finland
    The reproduction of acoustics is one of the key challenges in spatial sound reproduction. In order to reach high levels of realism and immersion, reproduced signals must be processed through a set of filters accounting for real-life phenomena. From the scattering and absorption of sound on walls to its diffraction around objects and our head, the propagation of sound waves in a room creates a complex sound field around a listener. While recreating all of the underlying parts of sound propagation is not yet within our reach for real-time applications, spatial sound techniques aim to minimize the computational cost by focusing the efforts on the most perceptually critical components. In this dissertation, the reproduction of reverberant sound fields is investigated through the development of analysis methods, reverberation algorithms, and decorrelation techniques. A particular emphasis is given to the perception and reproduction of directional characteristics present in late reverberation. Most reverberation algorithms consider that the sound energy is evenly distributed across space in all directions, after an initial period of time, since the diffusion of energy leads to more homogeneous and isotropic sound fields. However, previous work has demonstrated that insufficient diffusion in a room leads to anisotropic, and directional, late reverberation. In this dissertation, a complete framework is proposed for the objective and subjective analysis of directional characteristics as well as a novel delay-network reverberation method capable of producing direction-dependent decay properties. The reverberator is also expanded to offer efficient frequency- and direction-dependent processing. The proposed algorithm contains all the required elements for the auralization of reverberant sound fields, which may be modulated in real-time to support six-degrees-of-freedom sound reproduction. The decorrelation of audio signals, which occurs naturally during the diffusion of energy in a sound field, is another important aspect of sound reproduction. This dissertation considers the use of velvet-noise sequences, a special type of sparse noise signals, as decorrelation filters and offers a method to optimize their characteristics. Velvet noise is also proposed for the reproduction of an existing impulse response using a small set of time- and frequency-dependent information, along with a reverberator using velvet noise to improve the echo density of a delay network reverberator. Overall, the results contained in this dissertation offer new insights into the perceptual ramifications of reverberant sound fields containing directional characteristics and their reproduction. The methods presented bring applications in the context of immersive sound reproduction, such as in virtual and augmented reality.
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    Equalizer Design for Sound Reproduction
    (Aalto University, 2020) Liski, Juho; Rämö, Jussi, Dr., Aalto University, Finland; Välimäki, Vesa, Prof., Aalto University, Finland; Signaalinkäsittelyn ja akustiikan laitos; Department of Signal Processing and Acoustics; Aalto Acoustics Lab; Sähkötekniikan korkeakoulu; School of Electrical Engineering; Välimäki, Vesa, Prof., Aalto University, Department of Signal Processing and Acoustics, Finland
    Digital graphic equalizers are widely used in various applications, such as music production and reproduction, to control the magnitude responses of signals. They are straightforward to operate, with the gain at each band being the only controllable parameter. Different band divisions and different equalizer structures are used based on the requirements of the application or the available hardware. However, the accurate design of graphic equalizers is not straightforward due to band leakage, i.e., the interaction of the neighboring band filters leading to approximation errors. This is especially problematic in applications utilizing automatic equalizer updates, such as the adaptive equalization of headphones or loudspeakers based on ambient noise. In these applications, the target curve must be realized accurately and efficiently. This thesis focuses on accurate graphic-equalizer design and its applications. The first part discusses different equalizer structures and proposes a novel parallel graphic equalizer with a delayed IIR part. Almost all discussed designs achieve an accuracy of ±1 dB with the help of band filters whose responses are exactly controlled at three points. A least-squares method jointly optimizes the filter gains and accounts for the band leakage. It is shown that the number of design points should be double the number of bands, the accuracy is increased with band filters having a symmetric shape up to the Nyquist frequency, and the design parameters must be optimized for each design separately. The second part of the thesis discusses equalizer applications related to headphones. The proposed algorithms apply adaptive filters to hear-through processing and to individualization of the headphone response. The former algorithm estimates the isolation of the headphones and controls the equalizer in order to achieve acoustic transparency despite changes in the headphone fit. The latter example applies a similar adaptive filter to estimate the headphone magnitude response, which is then equalized to a selected target. Measurements indicate that both the magnitude-response estimation and equalization work accurately. The final part of the thesis focuses on loudspeaker equalization, and both the magnitude-only and phase equalization are elaborated. Multiple flat-panel loudspeakers are measured, and their sound is improved using a single-point equalizer design. A simulated loudspeaker is group-delay equalized with a frequency-sampled FIR filter, and a limited-band equalizer is shown to be supe- rior to the full-band group-delay equalization. The proposed accurate equalizers are suitable for a wide range of audio system equalization applications, both human- and algorithm-operated ones. The different structures enable the equalizers to be run on different hardware, and the low complexity of the equalizers comprising a single second-order section per band makes them especially suitable for mobile implementations.
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    Nonlinear abstract sound synthesis algorithms
    (Aalto University, 2013) Kleimola, Jari; Välimäki, Vesa, Prof., Aalto University, Finland; Signaalinkäsittelyn ja akustiikan laitos; Department of Signal Processing and Acoustics; Sähkötekniikan korkeakoulu; School of Electrical Engineering; Välimäki, Vesa, Prof., Aalto University, Finland
    This thesis explores abstract sound synthesis methods in digital musical instrument applications and proposes new algorithms for sound production, aliasing reduction, and efficient control of synthesis parameters. The specific focus is on nonlinear distortion and audio-rate modulation techniques, which are approached from two complementary perspectives. First, the classic view, built on closed-form mathematical expressions and computer algorithms, was seen to converge into a compound model where different abstract synthesis methods both generalize and reinforce each other. In this view, the recent phaseshaping technique was investigated as pipelined phaseshaper expressions, with applications in efficient and novel oscillator effects algorithms discovered in the thesis, such as an efficient super sawtooth simulation. In addition, a two-dimensional multi-point extension of the phase distortion method called vector phaseshaping synthesis (VPS) was proposed and demonstrated as an intuitive parametrization of the complex phase modulation technique. The method is well suited for contemporary multi-touch interaction and planar control- and audio-rate modulation. The second perspective into the nonlinear distortion and audio-rate modulation techniques, based on periodically linear time-varying filters, led to the discovery of a synthesis algorithm where the coefficients of an allpass filter chain are modulated at an audio rate. In addition, the filter approach enabled an alternative interpretation of the feedback amplitude modulation (FBAM) technique, whose first-order form was extended and a second-order form was introduced.To complement the sound production stage of digital musical instrument applications, two aliasing reduction methods were introduced, one based on scaled sinusoids and another on polynomial transition regions (PTR). The latter is currently the most efficient method for implementing alias-suppressed virtual analog oscillators. Finally, a streamlined control protocol that dramatically reduces the bandwidth of control data streams was proposed. The efficient and novel algorithms introduced in the thesis are useful for sound synthesis in resource constrained mobile platforms, web browsers, and in applications requiring a high polyphony.
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    Room Reverberation Prediction and Synthesis
    (Aalto University, 2022) Prawda, Karolina Anna; Schlecht, Sebastian J., Prof., Aalto University, Finland; Välimäki, Vesa, Prof., Aalto University, Finland; Signaalinkäsittelyn ja akustiikan laitos; Department of Signal Processing and Acoustics; Audio Signal Processing; Sähkötekniikan korkeakoulu; School of Electrical Engineering; Välimäki, Vesa, Prof., Aalto University, Department of Signal Processing and Acoustics, Finland
    In this dissertation, the discussion is centered around the sound energy decay in enclosed spaces. The works starts with the methods to predict the reverberation parameters, followed by the room impulse response measurement procedures, and ends with an analysis of techniques to digitally reproduce the sound decay. The research on the reverberation in physical spaces was initiated when the first formula to calculate room's reverberation time emerged. Since then, finding an accurate and reliable method to predict reverberation has been an important area of acoustic research. This thesis presents a comprehensive comparison of the most commonly used reverberation time formulas, describes their applicability in various scenarios, and discusses their accuracy when compared to results of measurements. The common sources of uncertainty in reverberation time calculations, such as bias introduced by air absorption and error in sound absorption coefficient, are analyzed as well. The thesis shows that decreasing such uncertainties leads to a good prediction accuracy of Sabine and Eyring equations in diverse conditions regarding sound absorption distribution. The measurement of the sound energy decay plays a crucial part in understanding the propagation of sound in physical spaces. Nowadays, numerous techniques to capture room impulse responses are available, each having its advantages and drawbacks. In this dissertation, the majority of commonly used measurement techniques are listed, whereas the exponential swept-sine is described in more detail. This work elaborates on the external factors that may impair the measurements and introduce error to their results, such as stationary and non-stationary noise, as well as time variance. The dissertation introduces Rule of Two, a method of detecting non-stationary disturbances in sweep measurements. It also shows the importance of using median as a robust estimator in non-stationary noise detection. Artificial reverberation is a popular sound effect, used to synthesize sound energy decay for the purpose of audio production. This dissertation offers an insight into artificial reverberation algorithms based on recursive structures. The filter design proposed in this work offers precise control over the decay rate while being efficient enough for real-time implementation. The thesis discusses the role of the delay lines and feedback matrix in achieving high echo density in feedback delay networks. It also shows that four velvet-noise sequences are sufficient to obtain smooth output in interleaved velvet noise reverberator. The thesis shows that the accuracy of reproduction increases the perceptual similarity between measured and synthesised impulse responses. The insights collected in this dissertation offer insights into the intricacies of reverberation prediction, measurement and synthesis. The results allow for reliable estimation of parameters related to sound energy decay, and offer an improvement in the field of artificial reverberation.