1. Embedded Sound Processing : Implementing the Echo Effect
Denist Oscar • Valendie Alix • Daniel Brescia • James Haralambides, PhD
Abstract
Embedding Signal Processing
Schematic Diagram for Circuit
Digitized audio is derived from sound reproduction of analog information by representing analog waveforms with a stream of digital bits. The main purpose of having a system with digital audio is to have the ability to store, retrieve and transmit signals without jeopardizing the loss of sound quality.
Digitization of audio signals allowed for the application of various effects that enhance the quality of the signal, eliminate undesired frequencies, or produce a better listening experience. Some of the most common effects include amplification, fading (in or out), noise removal, inverting, echoing, etc. In this project, we will generate a parameterized echo effect using programmable logic. Hardware implementations of signal processing algorithms are characterized by increased flexibility and an increased performance over software designs.
The echo effect is produced via repetition of the original signal after a delay and a decay factor have been applied to it. The delay parameter defines the amount of time that elapses before copies of the signal are repeated. The decay factor reduces the amplitude of the repeated signal until it eventually dies out.
The effect will be hardwired on a Spartan 3E board. The audio file will be stored in the core memory of the board as a collection of amplitude samples. The embedded design will make use of a memory reader module, a digital-to-analog converter, a controller module, and input and output interfaces to facilitate parameterization of the effect and sound reproduction. The memory reader is responsible for the retrieval of stored digital information while the controller combines audio samples representing different timing intervals to generate the echo effect. The job of the digital-to-analog converter is self-explanatory; it converts the signal to its corresponding analog form and forwards it to a computer speaker for reproduction.
Utilizing embedded system designs allows for increased operational performance of sound processing applications. Use of hardwired, reprogrammable designs allows users to easily reconfigure the design and its functionality partially or in its entirety more efficiently. An example of an embedded signal processing design is a digital signal processor (DSP), a specialized microprocessor with an optimized architecture for the fast operational needs of digital signal processing applications. This processor will utilize an algorithm that will constantly and rapidly process and convert audio signals from analog form to digital form.
Audio Input +
The principal components of the echoing effect embedded system: a) The clock module, b) the reader/controller, c) the DAC module, and d) the memory module.
An audio signal represented in the time domain. Amplitude range = [-1, 1]. Number of samples =
System Components and Operation
Audio files used for this project are stored in the .wav format (raw, uncompressed form). Files are read using Octave, a software application primarily used for numerical computations. Data retrieved are fractional integers in the range [-1, 1] and they represent the amplitudes of data samples recorded according to the sampling frequency used in the digitization process. To conform with requirements set by the DAC (digital-to-analog converter) module used in the design, data is converted to 12-bit unsigned binary integers. These samples are stored in the ROM memory component provided by the Spartan board.
The system operates in synchronous sequential mode; the clock is used to synchronize the various components so that sound is reproduced accurately.
The memory component is a single port ROM block that is programmed to store the audio samples of the input file. Memory data are read at a frequency that matches the sampling frequency of the raw data. For example, a sampling frequency of 8 kHz means that the clock speed of the main board (50 MHz) is slowed down by a factor of 6,250.
The DAC component receives 12-bit binary integers and converts them into a serial 32-bit stream that is forwarded to its output ports. The serialization of its input causes a delay factor that needs to be compensated by adjusting the clock speed of the component by a factor of 32 (the serialization factor).
The key component to the system is the reader module. The reader module is in direct synchronization with the memory module so that raw data are processed at rates equal to the sampling frequency. The echo effect is implemented using two parameters: decay and delay. To implement the decay parameter the reader module shifts data read from memory to the right by a number of positions equal to the decay value. A right binary shift is equivalent to division by a power of two. This results in a signal whose amplitude is diminished by the same factor which in turn produces lower voltage output levels for the signal. The amplitude reduction process is repeated at greater decay levels giving a sense of a gradually dying out audio signal. The delay parameter is implemented by simply deactivating memory reads for a duration specified by the delay value. The two parameters are combined to generate the full echo effect.
Introduction
What are Audio Effects?
Audio or sound effects are enhanced sound processes used to emphasize artistic or other content of audio signals. The term often refers to a process applied to a recording, without necessarily referring to the recording itself. Some sample sound effects include: amplification, reverberation, echo, bass boost, reversing, noise removal, fade in, fade out, etc.
The Echo Filter
The echo filter which is a simple delay line, repeats the audio a user has selected over and over again, with an optional reduction (decay) of the signal’s strength each time it is repeated. In between each repeat, a fixed time delay occurs.
Converting raw audio samples to 12-bit binary values.