Implementing Professional Audio Effects with DSPs

EURASIP Journal on Advances in Signal Processing, 2010

work on geometric wavefield decomposition which accounts for propagation phenomena such as diffusion and diffraction and serves as a computational engine for both wavefield rendering and binaural rendering. Still in the area of binaural rendering are the two contributions to this special issue, the first of which is by L. Wang et al., which addresses the longdebated problem of cross-talk cancellation. This paper is followed by that of M. Cobos et al., which proposes a method that allows us to avoid using a dummy head in binaural recording sessions.

In the field of musicians and recording engineers audio effects are mainly described and indicated by their acoustical effect. Audio effects can also be categorized from a technical point of view. The main criterion is found to be the type of modulation technique used to achieve the effect. After a short introduction to the different modulation types, three more sophisticated audio effect applications are presented, namely single sideband domain vibrato (mechanical vibrato bar simulation), a rotary speaker simulation, and an enhanced pitch transposing scheme.

Wiley Encyclopedia of Computer Science and Engineering, 2009

Innovations in Emerging Technology …, 2011

This paper shows the implementation of the echo and reverberation effects using the TI's C6713 DSK. The effects are simulated using Simulink. Those simulation models are used to generate the DSP code for the real-time implementation. Echo and Reverberation are two of the simplest applications of Digital Signal Processing. Artificial Reverberation is one of the most interesting DSP applications in music preparation. If the individual channels are simply added together, the resulting piece sounds frail and diluted, much as if the musicians were playing outdoors. This is because listeners are greatly influenced by the echo or reverberation content of the music, which is usually minimized in the sound studio. DSP allows artificial echoes and reverberation to be added during mix down to simulate various ideal listening environments. Echoes with delays of a few hundred milliseconds give the impression of cathedral like locations. Adding echoes with delays of 10-20 milliseconds provide the perception of more modest size listening rooms.

Sound synthesis involves creating a desired sound using software or algorithms and analysing the digital signal processing involved in the creation of the sound, rather than recording it. However, synthesis techniques are often too computationally complex for use in many scenarios. This project aims to implement and assess sound synthesis models with dynamic Level of Audio Detail (LOAD). The manipulations consist of modifying existing models to achieve a more or less complex implementation while still retaining the perceptual characteristics of the sound. The models implemented consist of sine waves, noise sources and filters that reproduce the desired sound, which could then be enhanced or reduced to provide dynamic LOAD. These different levels were then analysed in the time-frequency domain, and computational time and floating point operations were assessed as a function of the LOAD.

Proceedings of the 43rd IEEE Midwest Symposium on Circuits and Systems (Cat.No.CH37144)

The design of an audio processor card for generating special sound effects is presented in this paper. This is designed as an add-on card that plugs directly into the ISA bus of a PC. The card uses a Motorola DSP processor, DSP56001, for audio signal processing. External SRAM modules are used for program and data storage. A codec chip is employed to handle the digital interfacing between the DSP processor and the analog audio world.

Nepal Journal of Science and Technology, 2015

This paper describes the theory and implementation of audio effects such as echo, distortion and pitch-shift in Field Programmable Gate Array (FPGA). At first the mathematical formulation for generation of such effects is explained and then the algorithm is described for its implementation in FPGA using Very high speed integrated circuit hardware descriptive language (VHDL). The digital system being designed, which is synthesizable and reconfigurable, offers a great flexibility and scalability in designing and prototyping in FPGAs. The system is divided into three HDL blocks, each for echo, distortion, and pitch-shift effect generation, which are multiplexed in order to share the common ADC and DAC. The audio effect generator designed in this paper was successfully implemented in Spartan-3E FPGA utilizing the resources available effectively. There has been tremendous research being carried out in the field of IP core. Efficient IP cores designed to carry out digital signal processing are implemented in every modern device using configurable logics. This trend hasn't yet been realized in Nepal. Through the design and implementation of audio effect generator, this paper also aims at bringing the field of IP core development to limelight among scholars of Nepal.

Idle Task block specifies one or more functions to execute as background tasks in the code generated for the model. The functions are created from the function-call subsystems to which the Idle Task block is connected. Another block, the DSP/BIOS Task blocks spawn free-running tasks as separate DSP/BIOS threads. The spawned task runs the function-call subsystem connected to its output . The paper has been organized as follows: Section II talks about the theory behind various audio effects that are implemented. Section III briefly describes the real time schedules pertaining to the paper and, Section IV elaborates the analysis behind real time stress generation audio effect implementation and illustrates all other designs. The COCOMO analysis has been done in section V, and finally conclusions are drawn in section VI.

2007

This paper summarizes the results of the ESPRIT project APLODSP. The goal of this application experiment is to develop adequate models for the simulation of the non-linear behavior of loudspeakers and to design a dedicated audio processor to reduce sound distortion. This involves the definition of a systematic design flow for anti-distortion audio processors, and the effective exploitment of CAD tools for the automatic implementation of the defined algorithms. The audio processor will be implemented with a DSP, using state-of-theart tools for simulation, validation and synthesis. DSP is the emerging low cost technology for audio processing, and in particular for car-audio systems. In fact, car manufacturers are planning to reduce the cables inside the car and to use a single cable to distribute the main signals multiplexed all around the vehicle. This transition to digital audio signal transmission will foster the use of active loudspeakers, equipped with dedicated digital audio processors. The audio processor, designed and tested within this ESPRIT project, can be seen as a first step in this direction. It shows how the loudspeaker distortion can be reduced by digital signal processing, and it exploits the versatility of digital designs in order to allow hardware re-use for different car models. It also allows a very fast redesign to fit many different purposes. A semi-automatic design flow for the design of anti-distortion audio processors is available, which synthesizes a dedicated audio processor for each loudspeaker model, once suitable loudspeaker parameters have been specified. I.