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A study was performed using a virtual environment to investigate the relative importance of spatial audio fidelity and video resolution on perceived audio-visual quality and immersion. Subjects wore a head-mounted display and headphones and were presented with a virtual environment featuring music and speech stimuli using three levels each of spatial audio quality and video resolution. Spatial audio was rendered monaurally, binaurally with head-tracking, and binaurally with head-tracking and room acoustic rendering. Video was rendered at resolutions of 0.5 megapixels per eye, 1.5 megapixels per eye, and 2.5 megapixels per eye. Results showed that both video resolution and spatial audio rendering had a statistically significant effect on both immersion and audio-visual quality. Most strikingly, the results showed that under the conditions that were tested in the experiment, the addition of room acoustic rendering to head-tracked binaural audio had the same improvement on immersion as increasing the video resolution five-fold, from 0.5 megapixels per eye to 2.5 megapixels per eye.

The article is devoted to the systematization of the main modern methods of using reverberation in sound engineering and methodological bases of its use in the educational process in higher education institutions. Scientific novelty is to clarify the classification and systematization of the main modern methods of using artificial reverberation, as well as methodological principles of mastering artificial reverberation in the process of training. The purpose of the article is to systematize the main modern methods of using reverberation in sound engineering. To achieve this goal, the following methods were used: system-analytical, art history and comparative-historical. The main results and conclusions of the research. The evolution of digital technologies in the field of sound engineering, the introduction of reverberation algorithms have greatly contributed to the expansion of the use of artificial reverberation in sound engineering. The article briefly describes "convolutional" reverberation as one that operates with the most modern reverberation algorithm. Currently, a variety of effective methods of artificial reverberation using has been developed in sound engineering, which can be used to implement various creative tasks. The article describes the methods of using artificial reverberation for sound perspective creating; for back-vocal processing; for delay processing; for "softening" the sound of individual parts of musical instruments; for different signals of the same type processing; for solo batch processing using two reverb effects. Some types of artificial reverberation were used at different stages of the music industry development. The current state of software development allows the use of several reverb algorithms simultaneously, making it possible to create original sounds. In the process of mastering of various methods of using artificial reverberation by higher education students, it is best to start their acquaintance with artificial reverberation from the simplest reverberation effects, which can be implemented in "analog" effects processor, in which only some parameters vary based on the manufacturer's algorithms. Further mastering of artificial reverberation should be carried out through software (in particular, free VST-plugins of artificial reverberation) used in digital audio workstations (DAW).

ln order to improve the limitation of the number of delay limitation the existing feedback delay network method and the inability to determine the sub-band reverberation time, this paper introduces a feedback delay network for artificial reverberation based on the Householder feedback matrix and Butterworth filler bank . The Householder feedback matrix can maximize the echo density, the choice of delay lines number is relatively free. The Butterworth filter bank is convenient for specifying the sub-band reverberation time. In the simulation experiment, The validity of The proposed method was verified by spectrogram, perceptual evaluation of speech quality, and subjective evaluation. [ABSTRACT FROM AUTHOR]

Since the introduction of feedforward–feedback comb allpass filters by Schroeder and Logan, its popularity has not diminished due to its computational efficiency and versatile applicability in artificial reverberation, decorrelation, and dispersive system design. In this work, we present an extension to the Schroeder allpass filter by introducing frequency-dependent feedforward and feedback gains while maintaining the allpass characteristic. By this, we directly improve upon the design of Dahl and Jot which exhibits a frequency-dependent absorption but does not preserve the allpass property. At the same time, we also improve upon Gerzon's allpass filter as our design is both less restrictive and computationally more efficient. We provide a complete derivation of the filter structure and its properties. Furthermore, we illustrate the usefulness of the structure by designing an allpass decorrelation filter with frequency-dependent decay characteristics. [ABSTRACT FROM AUTHOR]

Feedback delay networks (FDNs) belong to a general class of recursive filters which are widely used in sound synthesis and physical modeling applications. We present a numerical technique to compute the modal decomposition of the FDN transfer function. The proposed pole finding algorithm is based on the Ehrlich-Aberth iteration for matrix polynomials and has improved computational performance of up to three orders of magnitude compared to a scalar polynomial root finder. The computational performance is further improved by bounds on the pole location and an approximate iteration step. We demonstrate how explicit knowledge of the FDN's modal behavior facilitates analysis and improvements for artificial reverberation. The statistical distribution of mode frequency and residue magnitudes demonstrate that relatively few modes contribute a large portion of impulse response energy. [ABSTRACT FROM AUTHOR]

Since the introduction of feedforward−feedback comb allpass filters by Schroeder and Logan, its popularity has not diminished due to its computational efficiency and versatile applicability in artificial reverberation, decorrelation, and dispersive system design. In this work, we present an extension to the Schroeder allpass filter by introducing frequency-dependent feedforward and feedback gains while maintaining the allpass characteristic. By this, we directly improve upon the design of Dahl and Jot which exhibits a frequency-dependent absorption but does not preserve the allpass property. At the same time, we also improve upon Gerzon’s allpass filter as our design is both less restrictive and computationally more efficient. We provide a complete derivation of the filter structure and its properties. Furthermore, we illustrate the usefulness of the structure by designing an allpass decorrelation filter with frequency-dependent decay characteristics.

Lossless Feedback Delay Networks (FDNs) are commonly used as a design prototype for artificial reverberation algorithms. The lossless property is dependent on the feedback matrix, which connects the output of a set of delays to their inputs, and the lengths of the delays. Both, unitary and triangular feedback matrices are known to constitute lossless FDNs, however, the most general class of lossless feedback matrices has not been identified. In this contribution, it is shown that the FDN is lossless for any set of delays, if all irreducible components of the feedback matrix are diagonally similar to a unitary matrix. The necessity of the generalized class of feedback matrices is demonstrated by examples of FDN designs proposed in literature. [ABSTRACT FROM PUBLISHER]

In this design study, we introduce a novel class of digital audio effects that extend the recently introduced modal processor approach to artificial reverberation and effects processing. These pitch and distortion processing effects mimic the design and sonics of a classic additive-synthesis-based electromechanical musical instrument, the Hammond tonewheel organ. As a reverb effect, the modal processor simulates a room response as the sum of resonant filter responses. This architecture provides precise, interactive control over the frequency, damping, and complex amplitude of each mode. Into this framework, we introduce two types of processing effects: pitch effects inspired by the Hammond organ's equal tempered "tonewheels", "drawbar" tone controls, vibrato/chorus circuit, and distortion effects inspired by the pseudo-sinusoidal shape of its tonewheels and electromagnetic pickup distortion. The result is an effects processor that imprints the Hammond organ's sonics onto any audio input. [ABSTRACT FROM AUTHOR]

In this design study, we introduce a novel class of digital audio effects that extend the recently introduced modal processor approach to artificial reverberation and effects processing. These pitch and distortion processing effects mimic the design and sonics of a classic additive-synthesis-based electromechanical musical instrument, the Hammond tonewheel organ. As a reverb effect, the modal processor simulates a room response as the sum of resonant filter responses. This architecture provides precise, interactive control over the frequency, damping, and complex amplitude of each mode. Into this framework, we introduce two types of processing effects: pitch effects inspired by the Hammond organ’s equal tempered “tonewheels”, “drawbar” tone controls, vibrato/chorus circuit, and distortion effects inspired by the pseudo-sinusoidal shape of its tonewheels and electromagnetic pickup distortion. The result is an effects processor that imprints the Hammond organ’s sonics onto any audio input.

The paper considers an artificial reverberation algorithm developed by Jot that simulates the behavior of a real concert hall and takes into account the reverberation time's variation with respect to frequency by using absorbent filters. This work includes both Jot's design of the absorbent filters and an improved version of it. Simulations are run and objective acoustic parameters are computed. The just noticeable difference (JND) is used to find the best artificial reverberator to ensure the best acoustic behavior. To assure the best acoustic behavior the just noticeable difference (JND) are used according to ISO standards. [ABSTRACT FROM AUTHOR]

The digital emulation of analog audio effects and synthesis components, through the simulation of lumped circuit components has seen a large amount of activity in recent years; electromechanical effects have seen rather less, primarily because they employ distributed mechanical components, which are not easily dealt with in a rigorous manner using typical audio processing constructs such as delay lines and digital filters. Spring reverberation is an example of such a system--a spring exhibits complex, highly dispersive behavior, including coupling between different types of wave propagation (longitudinal and transverse). Standard numerical techniques, such as finite difference schemes are a good match to such a problem, but require specialized design and analysis techniques in the context of audio processing. A model of helical spring vibration is introduced, along with a family of finite difference schemes suitable for time domain simulation. Various topics are covered, including numerical stability conditions, tuning of the scheme to the response of the model system, numerical boundary conditions and connection to an excitation and readout, implementation details, as well as computational requirements. Simulation results are presented, and full energy-based stability analysis appears in an Appendix. [ABSTRACT FROM AUTHOR]