Download Graphic Equalizer Design Using Higher-Order Recursive Filters
A straight-forward design of graphic equalizers with minimumphase behavior based on recently developed higher-order bandshelving filters is presented. Due to the high filter order, the gain in one band is almost completely independent from the gain in the other bands. Although no special care will be taken to design filters with complementary edges except for a suitable definition of the cut-off frequencies, the resulting amplitude deviation in the transitional region between the bands will be sufficiently low for many applications.
Download Delay-free audio coding based on ADPCM and error feedback
Real-time bidirectional audio applications, like microphones and monitor speakers in live performances, typically require communication systems with minimum latency. When digital transmission with limited bit rate is desired, this poses tight constraints on the algorithmic delay of the audio coding scheme. We present a delay-free approach employing adaptive differential pulse code modulation (ADPCM) and adaptive spectral shaping of the coding noise. To achieve zero-delay operation, both prediction and quantization logic of the ADPCM structure are realized in a backwardadaptive fashion. Noise shaping is accomplished via two feedback loops around the quantizer for efficient exploitation of the auditory selectivity and masking phenomena, respectively. Due to automatic optimization of the involved parameters, the performance of the proposed system is on par with that of prior low-delay approaches.
Download Automated Equalization for Room Resonance Suppression
Estimating room resonances in locations of big events and looking for counter-measures are normally done by sound engineers, mainly before the beginning of the event. In this paper an automation to enhance the audio quality in event rooms by suppressing the room resonances with a parametric equalizer of several high-Q peak filters is proposed. The room characteristics can be identified with few measurements in the listening area during the event, without applying an additional measuring signal (using its original sound signal). Based on this room characteristics the equalization filters are automatically designed. The results of several rooms tested with the automated equalization for room resonance suppression are presented as well as a discussion on the covered topics.
Download The Influence of Small Variations in a Simplified Guitar Amplifier Model
A strongly simplified guitar amplifier model, consisting of four stages, is presented. The exponential sweep technique is used to measure the frequency dependent harmonic spectra. The influence of small variations of the system parameters on the harmonic components is analyzed. The differences of the spectra are explained and visualized.
Download Impulse Response Measurement Techniques and their Applicability in the Real World
Measurement of impulse responses is a common task in audio signal processing. In this paper three common measurement techniques are reviewed: Maximum length sequences, exponentially swept sines and time delay spectrometry. The aim is to give the reader a brief tutorial of the methods with a special focus on deficiencies of the algorithms, aiding in the choice of the best algorithm for a task at hand. Additionally, for time delay spectrometry, a novel improvement is presented, lifting its restriction to relatively short impulse responses.
Download Discretization of Parametric Analog Circuits for Real-Time Simulations
The real-time simulation of analog circuits by digital systems becomes problematic when parametric components like potentiometers are involved. In this case the coefficients defining the digital system will change and have to be adapted. One common solution is to recalculate the coefficients in real-time, a possibly computationally expensive operation. With a view to the simulation using state-space representations, two parametric subcircuits found in typical guitar amplifiers are analyzed, namely the tone stack, a linear passive network used as simple equalizer and a distorting preamplifier, limiting the signal amplitude with LEDs. Solutions using trapezoidal rule discretization are presented and discussed. It is shown, that the computational costs in case of recalculation of the coefficients are reduced compared to the related DK-method, due to minimized matrix formulations. The simulation results are compared to reference data and show good match.
Download Physical Modelling of a Wah-wah Effect Pedal as a Case Study for Application of the Nodal DK Method to Circuits with Variable Parts
The nodal DK method is a systematic way to derive a non-linear state-space system as a physical model for an electrical circuit. Unfortunately, calculating the system coefficients requires inversion of a relatively large matrix. This becomes a problem when the system changes over time, requiring continuous recomputation of the coefficients. In this paper, we present an extension of the DK method to more efficiently handle variable circuit elements. The method is exemplified with the Dunlop Crybaby wah-wah effect pedal, as the continuous change of the potentiometer position is an extremely important aspect of the wah-wah effect.
Download Improved PVSOLA Time Stretching and Pitch Shifting for Polyphonic Audio
An advanced phase vocoder technique for high quality audio pitch shifting and time stretching is described. Its main concept is based on the PVSOLA time stretching algorithm which is already known to give good results on monophonic speech. Some enhancements are proposed to add the ability to process polyphonic material at equal quality by distinguishing between sinusoidal and noisy frequency components. Furthermore, the latency is reduced to get closer to a real time implementation. The new algorithm is embedded into a flexible pitch shifting and time stretching framework by adding transient detection and resampling. A subjective listening test is used to evaluate the new algorithm and to verify the improvements.
Download Simulation of Fender Type Guitar Preamp using Approximation and State-Space Model
This paper deals with usage of approximations for simulation of more complex audio circuits. A Fender type guitar preamp was chosen as a case study. This circuit contains two tubes and thus four nonlinear functions as well as it is a parametric circuit because of an integrated tone stack. A state-space approach was used for simulation and further, precomputed solution is approximated using nonuniform cubic splines.
Download Nonlinear-Phase Basis Functions in Quasi-Bandlimited Oscillator Algorithms
Virtual analog synthesis requires bandlimited source signal algorithms. An efficient methodology for the task expresses the traditionally used source waveforms or their time-derivatives as a sequence of bandlimited impulses or step functions. Approximations of the ideal bandlimited functions used in these quasi-bandlimited oscillator algorithms are typically linear-phase functions. In this paper, a general nonlinear-phase approach to the task is proposed. The discussed technique transforms an analog prototype filter to a digital filter using a modified impulse invariance transformation method that enables the impulse response to be sampled with arbitrary sub-sample shifts. The resulting digital filter is a set of parallel first- and/or second-order filters that are excited with short burst-like signals that depend on the offset of the waveform discontinuities. The discussed approach is exemplified with a number of design cases, illustrating different trade-offs between good alias reduction and low computational cost.