AbstractThursday May 17th 2007 TA: Amir Hirsch Author I: Dimitri Podoliev Author II: Will Buttinger MASSACHUSETTS INSTITUTE OF TECHNOLOGY Department of Electrical Engineering and Computer Science 6.111 – Introductory Digital Systems Laboratory FINAL PROJECT DIGITAL AUDIO PROCESSOR Abstract A digital electronic system that processes audio signals was implemented using verilog on an FPGA labkit. The system is built from a number of individual modules that each perform a different audio effect, based on input control parameters. In particular the effects implemented were a variable gain signal mixer, an audio delay with feedback control, low pass, high pass and band pass frequency filters, and an audio compressor. Audio is passed in and out of the system by an AC97 analog-to-digital converter. The connection between these modules and any additional external inputs is controlled by a highly configurable routing interconnect architecture, which can be configured from a computer via RS232 serial cable with simple commands, or from a VGA visual interface coming directly from the unit. Using the router, the modules were interconnected in different ways, with each configuration providing a different total audio modification. The interconnect technology that was developed is highly generic and extendable, and can easily be used as a miniature digital lab kit to construct simple circuits. It can be used in any situation where it is desirable to allow a user to interconnect pre-defined modules without having to wait for hardware synthesis.6.111 Introductory Digital Systems Laboratory Final Project – Digital Audio Processor Table of Contents INTRODUCTION 3 AC97 INTERFACE 4 FILTERS 5 Digital Signal Processing Analysis 5 Generic Architecture and Design 6 Low-Pass Filter 7 High-Pass Filter 7 Band-Pass Filter 8 GRAPHICAL USER INTERFACE 8 DCM 9 VGA Controller 9 Mouse Controller 9 Debounce Module 9 Display Module 9 Item Module 10 Intersection Module 10 TESTING AND DEBUGGING 15 26.111 Introductory Digital Systems Laboratory Final Project – Digital Audio Processor Introduction The processing of audio signals is a fundamental part of the music and entertainment industry. Although advanced audio-processing systems already exist, they cost tends to restrict their use to professional recording and editing studios. For the amateur instrumentalist, particularly the guitarist, who wishes to alter the way their instrument sounds, many individual effects units must be purchased and physically wired together. Some multi-effect processors exist, but these are expensive and none of these set-ups allow for true flexibility and creativity in the audio modification process. Also for the general music enthusiast, a lot of money can end up being spent on an amplifier that can perform the signal equalisations that may be desired. Because of these reasons, it is clear that there is a need for a highly configurable and customisable audio processing unit, which is contained within a single package and will allow the user to configure, control, and combine built-in audio effects in whatever way they wish. The system was implemented in verilog, and synthesised on to an FPGA labkit. By using this development environment, a new module can be created for each distinct audio effect that was to be included in the system, and then instantiated as many times as desired. Each module has input and output audio signals, as well as parameter inputs which control the exact behaviour of that module instance. For this particular body of work, the modules that were implemented were a mixer, which combines audio signals, a delay with feedback control and various frequency filters. The labkit’s built-in AC97 unit was used to convert an analog audio signal in to an 18-bit signed digital signal, and convert the final output digital signal in to an analog audio signal. The system would also make use of the labkit’s buttons and switches, which could be used as a demonstration for the setting of module parameters. It is key part of the system that the connection between these modules, inputs, and outputs be specified by the user. To allow the user this control, an interconnect architecture was developed that could be configured using simple commands sent from a computer via an RS232 serial cable, or by using a VGA interface with a mouse to point and click on the connections that are to be made. The basic overall block diagram for this system is shown in figure 1. Interconnect Architecture AC97 (built-in to labkit) AC97 controller VGA Interface RS232 Serial connection to computer Audio Processing Modules Control inputs (buttons, switches etc) analog_out 18-bit digitial audio sample Ac97_synch Signal_in Signal_out …. Vga interface signals Vga output Mouse input Serial configuration signal input analog_in Figure 1 – General block diagram for system 36.111 Introductory Digital Systems Laboratory Final Project – Digital Audio Processor The specific details of the modules within the system can be found in Section 2. The implementation and any simulated waveforms for each of these modules is given in Section 3, including any state transition diagrams. A discussion of how the system was tested and debugged is given in Section 4. A brief conclusion for the project is given in Section 5, where improvements and extensions to the project are discussed. In order to interact with music, it was essential to use the AC97 audio codec available on the labkit. This codec facilitated the ADC (Analog-to-Digital Conversion) and DAC (Digital-to-Analog Conversion) between the analog signal received from any music source, such as an IPod or a microphone, and the processed digital signal sent back to the speakers or headset. Each of the effect’s modules was unique in terms of the effect they have on the signal. The low-pass filter was designed to filter out the high frequency noise and to amplify the base and other low frequency sounds present in a given piece of music. The high-pass filter has exactly the opposite effect on the signal; it attenuates down the level of base, and amplifies the higher frequency sounds, such as the singer’s voice. The band-pass filter on the other hand, incorporates both of the effects described above; it highlights the intermediate frequency range of the music. The compressor is only effect that is rarely applied to already made music. Its
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