Jussi Pekonen - Academia.edu (original) (raw)

Papers by Jussi Pekonen

This paper investigates the audibility threshold of aliasing in computer-generated sawtooth signa... more This paper investigates the audibility threshold of aliasing in computer-generated sawtooth signals. Listening tests were conducted to find out how much the aliased frequency components below and above the fundamental must be attenuated for them to be inaudible. The tested tones comprised the fundamental frequencies 415, 932, 1480, 2093, 3136, and 3951 Hz, presented at 60-dB SPL and 44.1-kHz sampling rate. The results indicate that above the fundamental the aliased components must be attenuated 0, 19, 26, 27, 32, and 41 dB for the corresponding fundamental frequencies, and below the fundamental the attenuation of 0, 3, 6, 11, 12, and 11 dB, respectively, is sufficient. The results imply that the frequency-masking phenomenon affects the perception of aliasing and that the masking effect is more prominent above the fundamental than below it. The A-weighted noise- to-mask ratio is proposed as a suitable quality measure for sawtooth signals containing aliasing. It was shown that the bandlimited impulse train, the differentiated parabolic waveform, and the fourth-order polynomial bandlimited step function synthesis algorithms are perceptually alias-free up to 1, 2, and 4 kHz, respectively. General design rules for antialiasing sawtooth oscillators are derived based on the results and on knowledge of level-dependence of masking.

Classical geometric waveforms used in virtual analog synthesis suffer from aliasing distortion wh... more Classical geometric waveforms used in virtual analog synthesis suffer from aliasing distortion when simple sampling is used. An efficient antialiasing technique is based on expressing the aveforms as a filtered sum of time-shifted approximately bandlimited polynomial-spline basis functions. It is shown that by optimizing the coefficients of the basis function so that the aliasing distortion is perceptually minimized, the alias-free bandwidth of classical waveforms can be expanded. With the best of the case examples given here, the generated impulse-train and sawtooth waveform are alias-free up to fundamental frequencies over 10 kHz when the sampling rate is 44.1 kHz.

Digital subtractive synthesis is a popular music synthesis method, which requires oscillators tha... more Digital subtractive synthesis is a popular music synthesis method, which requires oscillators that are aliasing-free in a perceptual sense. It is a research challenge to find computationally efficient waveform generation algorithms that produce similar-sounding signals to analog music synthesizers but which are free from audible aliasing. A technique for approximately bandlimited waveform generation is considered that is based on a polynomial correction function, which is defined as the difference of a non-bandlimited step function and a polynomial approximation of the ideal bandlimited step function. It is shown that the ideal bandlimited step function is equivalent to the sine integral, and that integrated polynomial interpolation methods can successfully approximate it. Integrated Lagrange interpolation and B-spline basis functions are considered for polynomial approximation. The polynomial correction function can be added onto samples around each discontinuity in a non-bandlimited waveform to suppress aliasing. Comparison against previously known methods shows that the proposed technique yields the best tradeoff between computational cost and sound quality. The superior method amongst those considered in this study is the integrated third-order B-spline correction function, which offers perceptually aliasing-free sawtooth emulation up to the fundamental frequency of 7.8 kHz at the sample rate of 44.1 kHz.

The Hammond organ is an early electronic musical instrument, which was popular in the 1960s and 1... more The Hammond organ is an early electronic musical instrument, which was popular in the 1960s and 1970s. This paper proposes computationally efficient models for the Hammond organ and its rotating speaker system, the Leslie. Organ tones are generated using additive synthesis with appropriate features, such as a typical fast attack and decay envelope for the weighted sum of the harmonics and a small amplitude modulation simulating the construction inaccuracies of tone wheels. The key click is realized by adding the sixth harmonic modulated by an additional envelope to the original organ tone. For the Leslie speaker modeling we propose a new approach, which is based on time-varying spectral delay filters producing the Doppler effect. The resulting virtual organ, which is conceptually easy, has a pleasing sound and is computationally efficient to implement.

In the 1960s and 1970s, most electrical (or analog) synthesizers were based on a sound synthesis ... more In the 1960s and 1970s, most electrical (or analog) synthesizers were based on a sound synthesis technique called subtractive synthesis, in which a spectrally rich source signal, typically a sawtooth or rectangular pulse wave, is filtered with a time-varying filter. In the 1980s and 1990s this synthesis technique was almost overtaken by other synthesis techniques but in the mid-1990s musicians started again to show interest in the ''warm`` sound of subtractive synthesis. To meet this interest, a Swedish company Clavia introduced in 1995 the NordLead synthesizer that used the subtractive sound synthesis but in digital domain. Furthermore, they introduced the term ''virtual analog`` to represent the digital simulation of the analog audio devices. Since then research on digital signal processing methods for subtractive sound synthesis has increased and more and more interest has been shown to the topic in the past few years. In fact, the last year (2010) was again record-breaking when the number of publication on this topic is considered. In this paper, the history of the publications of virtual analog synthesis research is reviewed. In addition to the publication count over the years, the viewpoints of this paper include the distribution of the methodologies presented in the publications.

Signaalinkäsittelyn ja akustiikan laitos, Aalto-yliopiston sähkötekniikan korkeakoulu PL 13000, 0... more Signaalinkäsittelyn ja akustiikan laitos, Aalto-yliopiston sähkötekniikan korkeakoulu PL 13000, 00076 AALTO Jussi.Pekonen@aalto.fi, Vesa.Valimaki@tkk.fi

Discrete-time modelling strategies of analogue Moog sawtooth oscillator waveforms are presented. ... more Discrete-time modelling strategies of analogue Moog sawtooth oscillator waveforms are presented. Two alternative approaches suitable for real-time implementation are proposed, one modelling the analogue waveform in time-domain using phase distortion synthesis and another matching the spectrum of an existing antialiasing sawtooth oscillator to the corresponding analogue spectrum using a first-order IIR post-equalising filter. A parameter estimation procedure for both approaches is explained and performed. Performance evaluation using polynomial fits for the estimated parameters is carried out and good matches between the model outputs and recorded waveforms are obtained. The best match of the tested algorithms is produced by the phase distortion model and by post-equalising the fourth-order B-spline bandlimited step function sawtooth oscillator.

This paper will consider wave (amplitude) and phase signal shaping techniques for the digital emu... more This paper will consider wave (amplitude) and phase signal shaping techniques for the digital emulation of distortion effect processing. We examine how to determine the Wave- and Phase-shaping functions with harmonic amplitude and phase data. Three distortion effects units are used to provide test data. The action of the Wave- and Phase-shaping functions derived for these effects is demonstrated with the assistance of a super-resolution frequency-domain analysis technique.

In quasi-bandlimited classical waveform oscillators, the aliasing distortion present in a trivial... more In quasi-bandlimited classical waveform oscillators, the aliasing distortion present in a trivially sampled waveform can be reduced in the digital domain by applying a tabulated correction function. This paper presents an approach that applies the correction function in the differentiated domain by synthesizing a bandlimited impulse train (BLIT) that is integrated to obtain the desired bandlimited waveform. The ideal correction function of the BLIT method is infinitely long and in practice needs to be windowed. In order to obtain a good alias-reduction performance, long tables are typically required. It is shown that when a short look-up table is used, a windowed ideal correction function does not provide the best performance in terms of minimizing aliasing audibility. Instead, audibly improved alias-reduction performance can be obtained using a look-up table that has a parametric control over the low-order generations of aliasing. Some practical parametric look-up table designs are discussed in this paper, and their use and alias-reduction performance are exemplified. The look-up table designs discussed in this paper providing the best alias-reduction performance are parametric window functions and least-squares optimized multi-band FIR filter designs.

Green Circuits and …, Jan 1, 2010

Trivially sampled geometric waveforms such as the rectangular pulse wave used in subtractive soun... more Trivially sampled geometric waveforms such as the rectangular pulse wave used in subtractive sound synthesis suffer from aliasing caused by the discontinuities in the waveform or its derivative. Several algorithms for the reduction of aliasing distortion have been suggested, providing either complete removal or great suppression of aliasing. Some antialiasing oscillators utilize variable fractional delay filters as an essential part of the algorithm. In this paper, these oscillators are reviewed with an emphasis on motivating the use of the fractional delay filters.

aes.org

Honeybee swarms are characterized by their buzzing sound which can be very impressive close to a ... more Honeybee swarms are characterized by their buzzing sound which can be very impressive close to a hive. We present two techniques for real-time sound synthesis of swarming honeybees in a 3D multichannel setting. Both techniques are based on a source-filter model using a sawtooth oscillator and an all-pole equalization filter. The synthesis is controlled by the motion of the swarm, which is modeled in two different ways: as a set of coupled individual bees or with a swarming algorithm. The synthesized sound can be spatialized using the location information generated by the model. The proposed methods are capable of producing a realistic honeybee swarm effect to be used in, e.g., virtual reality applications.

Center for Computer …, Jan 1, 2009

A recently introduced structure to implement a continuously smooth spectral delay, based on a cas... more A recently introduced structure to implement a continuously smooth spectral delay, based on a cascade of first-order allpass filters and an equalizing filter, is described and the properties of this spectral delay filter are reviewed. A new amplitude envelope equalizing filter for the spectral delay filter is proposed and the properties of structures utilizing feedback and/or time-varying filter coefficients are discussed. In addition, the stability conditions for the feedback and the time-varying structures are derived. A spectral delay filter can be used for synthesizing chirp-like sounds or for modifying the timbre of arbitrary audio signals. Sound examples on the use of the spectral delay filters utilizing the structures discussed in this paper can be found at http://www.acoustics.hut.fi/publications/papers/dafx09-sdf/.

DAFx 09 proceedings of …, Jan 1, 2009

This article discusses Phase Distortion synthesis and its application to arbitrary input signals.... more This article discusses Phase Distortion synthesis and its application to arbitrary input signals. The main elements that compose the technique are presented. Its similarities to Phase Modulation are discussed and the equivalence between the two techniques is explored. Two alternative methods of distorting the phase of an arbitrary signal are presented. The first is based on the audio-rate modulation of a first-order allpass filter coefficient. The other method relies on a re-casting of the Phase Modulation equation, which leads to a heterodyned form of waveshaping. The relationship of these implementations to the original technique is explored in detail. Complementing the article, a number of examples are discussed, demonstrating the application of the technique as an interesting digital audio effect.

Proceedings of the …

This paper describes a sound synthesis technique that modulates the coefficients of allpass filte... more This paper describes a sound synthesis technique that modulates the coefficients of allpass filter chains using audio-rate frequencies. It was found that modulating a single allpass filter section produces a feedback AM--like spectrum, and that its bandwidth is extended and further processed by non-sinusoidal FM when the sections are cascaded. The cascade length parameter provides dynamic bandwidth control to prevent upper range aliasing artifacts, and the amount of spectral content within that band can be controlled using a modulation index parameter. The technique is capable of synthesizing rich and evolving timbres, including those resembling classic virtual analog waveforms. It can also be used as an audio effect with pitch-tracked input sources. Soft- ware and sound examples are available at http://www.acoustics.hut.fi/publications/papers/dafx09-cm/.

aes.org

An essential component of digital emulations of subtractive synthesizer systems are the algorithm... more An essential component of digital emulations of subtractive synthesizer systems are the algorithms used to generate the classic oscillator waveforms of sawtooth, square and triangle waves. Not only should these be perceived to be authentic sonically, but they should also exhibit minimal aliasing distortions and be computationally efficient to implement. This paper examines a set of novel techniques for the production of the classic oscillator waveforms of Analogue subtractive synthesis that are derived from using amplitude or phase distortion of a mono-component input waveform. Expressions for the outputs of these distortion methods are given that allow parameter control to ensure proper bandlimited behavior. Additionally, their implementation is demonstrably efficient. Lastly, the results presented illustrate their equivalence to their original Analogue counterparts.

126th AES Convention, …, Jan 1, 2009

In multi-channel reproduction of spatial audio with first-order Ambisonics the loudspeaker signal... more In multi-channel reproduction of spatial audio with first-order Ambisonics the loudspeaker signals are relatively coherent, which produces prominent coloration. The coloration artifacts have been suggested to depend on the acoustics of the listening room. This dependency was researched with subjective listening tests in an anechoic chamber with an octagonal loudspeaker setup. Different virtual listening rooms were created by adding diffuse reverberation with 0.25 seconds RT60 using a 3D 16-channel loudspeaker setup. In the test, the subjects compared the audio quality in the virtual rooms. The results suggest that optimal audio quality was obtained when the virtual room effect and the direct sound were on equal level at the listening position.

Acoustics, Speech and …

This paper examines a recently introduced technique for sound synthesis that uses a coefficient m... more This paper examines a recently introduced technique for sound synthesis that uses a coefficient modulated allpass filter to cause phase modifications to its input signal. The intention in this work is to outline some of the properties of the coefficient modulated allpass filter and then to establish a connection between this new method and the older technique of phase distortion. Results are presented to demonstrate how the allpass technique provides a spectrally richer output signal.

This paper investigates the audibility threshold of aliasing in computer-generated sawtooth signa... more This paper investigates the audibility threshold of aliasing in computer-generated sawtooth signals. Listening tests were conducted to find out how much the aliased frequency components below and above the fundamental must be attenuated for them to be inaudible. The tested tones comprised the fundamental frequencies 415, 932, 1480, 2093, 3136, and 3951 Hz, presented at 60-dB SPL and 44.1-kHz sampling rate. The results indicate that above the fundamental the aliased components must be attenuated 0, 19, 26, 27, 32, and 41 dB for the corresponding fundamental frequencies, and below the fundamental the attenuation of 0, 3, 6, 11, 12, and 11 dB, respectively, is sufficient. The results imply that the frequency-masking phenomenon affects the perception of aliasing and that the masking effect is more prominent above the fundamental than below it. The A-weighted noise- to-mask ratio is proposed as a suitable quality measure for sawtooth signals containing aliasing. It was shown that the bandlimited impulse train, the differentiated parabolic waveform, and the fourth-order polynomial bandlimited step function synthesis algorithms are perceptually alias-free up to 1, 2, and 4 kHz, respectively. General design rules for antialiasing sawtooth oscillators are derived based on the results and on knowledge of level-dependence of masking.

Classical geometric waveforms used in virtual analog synthesis suffer from aliasing distortion wh... more Classical geometric waveforms used in virtual analog synthesis suffer from aliasing distortion when simple sampling is used. An efficient antialiasing technique is based on expressing the aveforms as a filtered sum of time-shifted approximately bandlimited polynomial-spline basis functions. It is shown that by optimizing the coefficients of the basis function so that the aliasing distortion is perceptually minimized, the alias-free bandwidth of classical waveforms can be expanded. With the best of the case examples given here, the generated impulse-train and sawtooth waveform are alias-free up to fundamental frequencies over 10 kHz when the sampling rate is 44.1 kHz.

Digital subtractive synthesis is a popular music synthesis method, which requires oscillators tha... more Digital subtractive synthesis is a popular music synthesis method, which requires oscillators that are aliasing-free in a perceptual sense. It is a research challenge to find computationally efficient waveform generation algorithms that produce similar-sounding signals to analog music synthesizers but which are free from audible aliasing. A technique for approximately bandlimited waveform generation is considered that is based on a polynomial correction function, which is defined as the difference of a non-bandlimited step function and a polynomial approximation of the ideal bandlimited step function. It is shown that the ideal bandlimited step function is equivalent to the sine integral, and that integrated polynomial interpolation methods can successfully approximate it. Integrated Lagrange interpolation and B-spline basis functions are considered for polynomial approximation. The polynomial correction function can be added onto samples around each discontinuity in a non-bandlimited waveform to suppress aliasing. Comparison against previously known methods shows that the proposed technique yields the best tradeoff between computational cost and sound quality. The superior method amongst those considered in this study is the integrated third-order B-spline correction function, which offers perceptually aliasing-free sawtooth emulation up to the fundamental frequency of 7.8 kHz at the sample rate of 44.1 kHz.

The Hammond organ is an early electronic musical instrument, which was popular in the 1960s and 1... more The Hammond organ is an early electronic musical instrument, which was popular in the 1960s and 1970s. This paper proposes computationally efficient models for the Hammond organ and its rotating speaker system, the Leslie. Organ tones are generated using additive synthesis with appropriate features, such as a typical fast attack and decay envelope for the weighted sum of the harmonics and a small amplitude modulation simulating the construction inaccuracies of tone wheels. The key click is realized by adding the sixth harmonic modulated by an additional envelope to the original organ tone. For the Leslie speaker modeling we propose a new approach, which is based on time-varying spectral delay filters producing the Doppler effect. The resulting virtual organ, which is conceptually easy, has a pleasing sound and is computationally efficient to implement.

In the 1960s and 1970s, most electrical (or analog) synthesizers were based on a sound synthesis ... more In the 1960s and 1970s, most electrical (or analog) synthesizers were based on a sound synthesis technique called subtractive synthesis, in which a spectrally rich source signal, typically a sawtooth or rectangular pulse wave, is filtered with a time-varying filter. In the 1980s and 1990s this synthesis technique was almost overtaken by other synthesis techniques but in the mid-1990s musicians started again to show interest in the ''warm`` sound of subtractive synthesis. To meet this interest, a Swedish company Clavia introduced in 1995 the NordLead synthesizer that used the subtractive sound synthesis but in digital domain. Furthermore, they introduced the term ''virtual analog`` to represent the digital simulation of the analog audio devices. Since then research on digital signal processing methods for subtractive sound synthesis has increased and more and more interest has been shown to the topic in the past few years. In fact, the last year (2010) was again record-breaking when the number of publication on this topic is considered. In this paper, the history of the publications of virtual analog synthesis research is reviewed. In addition to the publication count over the years, the viewpoints of this paper include the distribution of the methodologies presented in the publications.

Signaalinkäsittelyn ja akustiikan laitos, Aalto-yliopiston sähkötekniikan korkeakoulu PL 13000, 0... more Signaalinkäsittelyn ja akustiikan laitos, Aalto-yliopiston sähkötekniikan korkeakoulu PL 13000, 00076 AALTO Jussi.Pekonen@aalto.fi, Vesa.Valimaki@tkk.fi

Discrete-time modelling strategies of analogue Moog sawtooth oscillator waveforms are presented. ... more Discrete-time modelling strategies of analogue Moog sawtooth oscillator waveforms are presented. Two alternative approaches suitable for real-time implementation are proposed, one modelling the analogue waveform in time-domain using phase distortion synthesis and another matching the spectrum of an existing antialiasing sawtooth oscillator to the corresponding analogue spectrum using a first-order IIR post-equalising filter. A parameter estimation procedure for both approaches is explained and performed. Performance evaluation using polynomial fits for the estimated parameters is carried out and good matches between the model outputs and recorded waveforms are obtained. The best match of the tested algorithms is produced by the phase distortion model and by post-equalising the fourth-order B-spline bandlimited step function sawtooth oscillator.

This paper will consider wave (amplitude) and phase signal shaping techniques for the digital emu... more This paper will consider wave (amplitude) and phase signal shaping techniques for the digital emulation of distortion effect processing. We examine how to determine the Wave- and Phase-shaping functions with harmonic amplitude and phase data. Three distortion effects units are used to provide test data. The action of the Wave- and Phase-shaping functions derived for these effects is demonstrated with the assistance of a super-resolution frequency-domain analysis technique.

In quasi-bandlimited classical waveform oscillators, the aliasing distortion present in a trivial... more In quasi-bandlimited classical waveform oscillators, the aliasing distortion present in a trivially sampled waveform can be reduced in the digital domain by applying a tabulated correction function. This paper presents an approach that applies the correction function in the differentiated domain by synthesizing a bandlimited impulse train (BLIT) that is integrated to obtain the desired bandlimited waveform. The ideal correction function of the BLIT method is infinitely long and in practice needs to be windowed. In order to obtain a good alias-reduction performance, long tables are typically required. It is shown that when a short look-up table is used, a windowed ideal correction function does not provide the best performance in terms of minimizing aliasing audibility. Instead, audibly improved alias-reduction performance can be obtained using a look-up table that has a parametric control over the low-order generations of aliasing. Some practical parametric look-up table designs are discussed in this paper, and their use and alias-reduction performance are exemplified. The look-up table designs discussed in this paper providing the best alias-reduction performance are parametric window functions and least-squares optimized multi-band FIR filter designs.

Green Circuits and …, Jan 1, 2010

Trivially sampled geometric waveforms such as the rectangular pulse wave used in subtractive soun... more Trivially sampled geometric waveforms such as the rectangular pulse wave used in subtractive sound synthesis suffer from aliasing caused by the discontinuities in the waveform or its derivative. Several algorithms for the reduction of aliasing distortion have been suggested, providing either complete removal or great suppression of aliasing. Some antialiasing oscillators utilize variable fractional delay filters as an essential part of the algorithm. In this paper, these oscillators are reviewed with an emphasis on motivating the use of the fractional delay filters.

aes.org

Honeybee swarms are characterized by their buzzing sound which can be very impressive close to a ... more Honeybee swarms are characterized by their buzzing sound which can be very impressive close to a hive. We present two techniques for real-time sound synthesis of swarming honeybees in a 3D multichannel setting. Both techniques are based on a source-filter model using a sawtooth oscillator and an all-pole equalization filter. The synthesis is controlled by the motion of the swarm, which is modeled in two different ways: as a set of coupled individual bees or with a swarming algorithm. The synthesized sound can be spatialized using the location information generated by the model. The proposed methods are capable of producing a realistic honeybee swarm effect to be used in, e.g., virtual reality applications.

Center for Computer …, Jan 1, 2009

A recently introduced structure to implement a continuously smooth spectral delay, based on a cas... more A recently introduced structure to implement a continuously smooth spectral delay, based on a cascade of first-order allpass filters and an equalizing filter, is described and the properties of this spectral delay filter are reviewed. A new amplitude envelope equalizing filter for the spectral delay filter is proposed and the properties of structures utilizing feedback and/or time-varying filter coefficients are discussed. In addition, the stability conditions for the feedback and the time-varying structures are derived. A spectral delay filter can be used for synthesizing chirp-like sounds or for modifying the timbre of arbitrary audio signals. Sound examples on the use of the spectral delay filters utilizing the structures discussed in this paper can be found at http://www.acoustics.hut.fi/publications/papers/dafx09-sdf/.

DAFx 09 proceedings of …, Jan 1, 2009

This article discusses Phase Distortion synthesis and its application to arbitrary input signals.... more This article discusses Phase Distortion synthesis and its application to arbitrary input signals. The main elements that compose the technique are presented. Its similarities to Phase Modulation are discussed and the equivalence between the two techniques is explored. Two alternative methods of distorting the phase of an arbitrary signal are presented. The first is based on the audio-rate modulation of a first-order allpass filter coefficient. The other method relies on a re-casting of the Phase Modulation equation, which leads to a heterodyned form of waveshaping. The relationship of these implementations to the original technique is explored in detail. Complementing the article, a number of examples are discussed, demonstrating the application of the technique as an interesting digital audio effect.

Proceedings of the …

This paper describes a sound synthesis technique that modulates the coefficients of allpass filte... more This paper describes a sound synthesis technique that modulates the coefficients of allpass filter chains using audio-rate frequencies. It was found that modulating a single allpass filter section produces a feedback AM--like spectrum, and that its bandwidth is extended and further processed by non-sinusoidal FM when the sections are cascaded. The cascade length parameter provides dynamic bandwidth control to prevent upper range aliasing artifacts, and the amount of spectral content within that band can be controlled using a modulation index parameter. The technique is capable of synthesizing rich and evolving timbres, including those resembling classic virtual analog waveforms. It can also be used as an audio effect with pitch-tracked input sources. Soft- ware and sound examples are available at http://www.acoustics.hut.fi/publications/papers/dafx09-cm/.

aes.org

An essential component of digital emulations of subtractive synthesizer systems are the algorithm... more An essential component of digital emulations of subtractive synthesizer systems are the algorithms used to generate the classic oscillator waveforms of sawtooth, square and triangle waves. Not only should these be perceived to be authentic sonically, but they should also exhibit minimal aliasing distortions and be computationally efficient to implement. This paper examines a set of novel techniques for the production of the classic oscillator waveforms of Analogue subtractive synthesis that are derived from using amplitude or phase distortion of a mono-component input waveform. Expressions for the outputs of these distortion methods are given that allow parameter control to ensure proper bandlimited behavior. Additionally, their implementation is demonstrably efficient. Lastly, the results presented illustrate their equivalence to their original Analogue counterparts.

126th AES Convention, …, Jan 1, 2009

In multi-channel reproduction of spatial audio with first-order Ambisonics the loudspeaker signal... more In multi-channel reproduction of spatial audio with first-order Ambisonics the loudspeaker signals are relatively coherent, which produces prominent coloration. The coloration artifacts have been suggested to depend on the acoustics of the listening room. This dependency was researched with subjective listening tests in an anechoic chamber with an octagonal loudspeaker setup. Different virtual listening rooms were created by adding diffuse reverberation with 0.25 seconds RT60 using a 3D 16-channel loudspeaker setup. In the test, the subjects compared the audio quality in the virtual rooms. The results suggest that optimal audio quality was obtained when the virtual room effect and the direct sound were on equal level at the listening position.

Acoustics, Speech and …

This paper examines a recently introduced technique for sound synthesis that uses a coefficient m... more This paper examines a recently introduced technique for sound synthesis that uses a coefficient modulated allpass filter to cause phase modifications to its input signal. The intention in this work is to outline some of the properties of the coefficient modulated allpass filter and then to establish a connection between this new method and the older technique of phase distortion. Results are presented to demonstrate how the allpass technique provides a spectrally richer output signal.