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 Time-Domain Chroma Extraction In this paper, a novel chroma extraction technique called TimeDomain Chroma Extraction (TDCE) is introduced. In comparison to many other known schemes, the calculation of a time-frequency representation is unnecessary since the TDCE is a pure sample-bysample technique. It mainly consists of a pitch tracking module that is implemented with a phase-locked loop (PLL). A set of 24 bandpass filters over two octaves is designed with the F 0 output of the pitch tracker to estimate a chroma vector. To verify the performance of the TDCE, a simple chord recognition algorithm is applied to the chroma output. The experimental results show that this novel time-domain chroma extraction technique yields good results while requiring only minor complexity and thus, enables the extraction of musical features in real-time on low-cost DSP platforms.
Download Comparison of Various Predictors for Audio Extrapolation In this study, receiver-based audio error concealment in the context of low-latency Audio over IP transmission is analyzed. Therefore, the well-known technique of audio extrapolation is investigated concerning its usability in real-time scenarios, its applied prediction techniques and various transmission parameters. A large-scale automated evaluation with PEAQ and a MUSHRA listening test reveal the performance of the various extrapolation setups. The results show the suitability of extrapolation to perform audio error concealment in real-time and the qualitative superiority of block based methods over sample based methods.