CAREER implemented in hardware using a VHDL




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Title: Digital Compensation
of Distortion in Audio Systems

Location: Department of Electrical Engineering,
Linkoping University, Sweden

Supervisor: Kent
Palmkvist, Pär Gunnars Risberg



Audio systems are usually found to cost an exorbitant amount, and
while cheaper loudspeakers and other audio systems are alternatives, they do
not provide the best quality. The accuracy of these less costly systems can be
improved by using real-time compensation, thus reducing the need for pricey
systems. The availability of hardware that can digitally compensate for
loudspeaker distortion and other characteristics, enables this.



The nature of
this project is investigatory since our main purpose is to thoroughly examine the
methods of digital compensation and evaluate all the available techniques of
reducing distortion in audio systems. Since distortions can cause the overall
performance of a loudspeaker to deteriorate, it is extremely necessary to
invest in new methods that can eliminate these unwanted effects such as distortions
and phase delays. In addition, louder bass sounds require more energy to be
expended but the precision of audio systems gets downgraded significantly in
doing so. This is the main issue that needs to be considered during this



The main
objective of this project was to eradicate distortions and other harmful consequences
of loudspeakers and amplifiers by using digital compensation methods and thereby
resulting in improved performance. This project was performed to acquire more
information about distortions, which caused by a high input volume so that various
methods can be developed for detecting it and consequently, compensating for it.
The major goal was to implement a fully working model that could be implemented
in hardware using a VHDL (VHSIC Hardware Description Language) module so that high
volume input sequences could be operated on. Such a system would result in the
modification of only those portions of the signal that cause distortions,
resulting in enhanced performance and accuracy while playing bass at a loud



All tasks associated
with the project were performed at Actiwave AB. These tasks were divided
amongst the team and include:


·   Detailed
examination of the audio system to find out how a too high input signal causing
perceptible distortions can be detected digitally.

·   Construct a model
to discover and compensate for distortions and understand how the model works
on removing them.

·   Realize the
developed model into a real-world, working hardware system that is consistent
with processing time and other hardware limitations.

·   Compare the
compensated data with the original distorted data and evaluate how well the
system performs.



This project
was performed by a two-member team and we were responsible for performing the different
activities required to complete the project. All the activities were overseen at
Actiwave AB by Pär Gunnars Risberg, who provided us with all the necessary equipment
and resources while the experiment was being carried out. Our supervisor at the
university, Kent Palmkvistr reviewed each stage of the project and ensured that
all the required tasks were performed as per planned.


completion of the project, the final report was compiled which was reviewed by our
supervisor, who then submitted it to the Head of the Department. The report was
then forwarded to the Education Council and Research Council for record











In this project, I was
responsible for carrying out the different examinations required to ascertain the
cause of distortions that can be heard in loudspeakers and other audio systems
caused by a high input signal. This process consisted of several steps:


Sequence Analysis

A small and reverberant room was
used in this experiment for playing, recording and studying an audio sequence to
find hints of distortion. A loudspeaker and amplifier were used and the
sequence was played at such a volume that the effects of the distortions could
be heard unmistakably. Certain obvious differences were observed upon comparing
the played audio sequence with the recorded one, mostly because the recorded
signal consists of the impulse responses of the room as well as that of the
audio equipment, which affects the resulting wave of the signal. The distortion
also causes higher frequencies to be added to the played sequence, however,
this data is not enough to provide a solid insight into the divergence of the
played signal from the recorded signal or to lead to any kind of conclusions
about the presence of distortions in the signal.


the Cause of Distortion

I replayed the recorded signal using
a loudspeaker and amplifier and an electronic test instrument called an
oscilloscope, was linked to the input of the loudspeaker to allow for an
inspection of altering signal voltages as a two-dimensional graph of the
signal(s). The distortion was seen to overlap with the output voltage of the amplifier,
thereby clipping the signal. It can thus be presumed that the incapability of
the loudspeaker to imitate high input signals does not cause distortions,
instead, the reason behind it is the failure of the amplifier to generate the
correct signals and forward them to the loudspeaker. I then exchanged this analogue
amplifier for another one that was produced by Actiwave themselves since they are
class-D amplifiers. The test was repeated with this amplifier, however, the
results observed did not change.


of Model

A model was constructed in
Simulink to understand how the distortions affect the rest of the signal.  I performed two kinds of test, one tone and two-tone
to identify the frequencies at which the distortion occurs and how saturation
of the signal affects its magnitude. After conducting these tests, it was
observed that the undesirable signals were occurring at a lower frequency than
the original one. I also noticed that the distortion could be detected without
much hassle by the human ear, whether it was caused by clipping of the signal
or not. In order to maintain high precision and accuracy, distortion in the
original signal must be controlled and kept to the absolute minimum.


Cause of Saturation

Voltage saturation can be caused
by a number of different sources, ranging from the structure of the amplifier (class-D
or analogue) to the inadequacies of the amplifier. These reasons are analysed
further by examining the relationship between frequency and voltage saturation.
A bass driver test was performed where sine waves were applied at different
frequencies and volumes and the minimum voltage for distortion that could be
heard, as well as the maximum voltage for non-discernable distortion, were
recorded. A loudspeaker, as well as an amplifier, were used, however, the
tweeter was disconnected while recording the observations. The same test is
then repeated after connecting the tweeter. The results of this second test indicate
that the output voltage is restricted by a lower input voltage signal and since
distortions couldn’t be heard when the tweeter wasn’t connected, the tweeter
actually acts as the source which gives audible distortions.



For this experiment, we are
only interested in the distortions caused by bass in audio systems, so I
decided to not study the distortion in the tweeter any further. The key to
reducing distortion is in limiting the output voltage (to 22V) to make sure
that the signal does not get clipped. One more method that could solve our
problem is to limit the bass in those channels that replicate high frequencies.



In order to ensure that our original
input signal does not get clipped, we decided to use a limiter to limit the
values reaching the available channels. The limiter is only applied at low
frequencies since that’s where most of the distortion was found during our
analysis. This was the basis for the proposed model of the solution that was
implemented in Simulink. The limiter employed for this purpose has the following
components or phases:


LP prefiltering, LP post-filtering and preserving full
frequency signal: An LP filter must be connected to the audio system to ensure that
only lower frequencies are used when limiting the bass, as well as removing perceptible
clipping sounds caused by amplitude changes.

Altering amplitude of the bass channel: The
amplitude of the bass channel is reduced using time multiplexing, scaling or
FFT and notch filters. After observing each method carefully, it was found that
scaling would work best for our required purpose.

Decision-making block: This block is responsible for
deciding whether the bass should be limited or not.



The model simulated in Simulink
was implemented by us using preexisting VHDL modules. This included adjusting the
biquad filter structure that had been provided to us by Actiwave so that it
could sequentially execute different sets of inputs using different filter
coefficients in the same hardware system. Scaling of the signal is done by
using an IP core multiplier and an FSM (Finite State Machine) and two frequency
bands are added and then instantiated by an IP core adder. The constructed
model accomplishes the completion of the following tasks:

Generation of a full range signal after applying
limitation of input voltage so that this methodology using limiters can be
implemented in various kinds of audio systems, with loudspeakers and amplifiers
having distinct frequencies within which they operate.

Weaken or limit the bass wherever it is required to
prevent distortion that can be heard and thus maintaining the accuracy and precision
of the sound.

Preventing any phase delays that can modify the detected

Making sure that the developed system is capable of
being updated and modified, should the need arise.



To allow for comparison of measurements
from different loudspeakers, Total Harmonic Distortion (THD) measurements are
usually taken at a certain Sound Pressure Level (SPL). These measures include
the THD of the amplifier as well as the distortion of the loudspeaker and the
results when the limiter is on or off can then be compared. This is the main
test because with the limiter if it performs its required job and works properly,
the signal should avoid being clipped and the output voltage would then be constrained
within a certain level. We made use of the freely available software ARTA,
which is a collection of programs that facilitate the measurement and analysis
of audio.


The collected data shows that the
limiter employed in the model limits the input voltage values and by limiting
the output voltage, reduces the distortion produced. It was also observed that the
measured THD becomes constant once it crosses a pre-specified threshold point,
thus averting clipping of the signal.



It is difficult to recognize
the signs of distortion from the playback of a recording, hence we had to rely
on human ears to pick up the distortion. The limitations of an amplifier cause saturation
of the input voltage. The amplifier circuit may also be shut down as a consequence
of too much sinking from it. The limiter has to be positioned before the volume
multiplicator, the volume needs to be controlled. However, this can neither be
implemented in Simulink nor in the VHDL hardware model. This is because, in
Simulink models, the simulations work at a fixed volume level and the limiter
does not work in real-time.


If we do not employ a limiter, one
might be able to increase the sound to as high a level as they want and a
crackling sound will be produced due to amplified distortion, thus resulting in
decreased precision and defeating the whole purpose of the experiment. When a
limiter is used, the unpleasant crackling can be avoided, however, the bass gets
restricted, meaning that the bass remains constant even when the volume is
turned up. This limited output can displease the listeners.



The conclusions obtained from
conducting this experiment is that a real-time implementation of the simulation
model described in the experiment can be created using low-cost hardware
modules. This execution identifies the frequencies at which the bass produces distortion
and performs functioning to compensate for it. The realised, operation model
improved the accuracy and performance of the signal even when there was a very
high input voltage, i.e, the sound was being played at a loud volume. It also
ensured that the hardware model was maintainable and effective. When a limiter was
used, the sounds from the loudspeaker seemed to be more controlled and all distortion
and its unwanted effects were removed.



The report for this project was
compiled by our two-member team by using various records created throughout the
duration of the project. Each phase in the execution of the project involved making
notes consisting of various analysis results, measured data or other related
information. Upon completion of the project, the final completed report was
submitted by us to the Head of the Department.



on this project allowed me to discover various methods of experimentation and
analysis. It also helped me strengthen my capabilities of performing tasks as
part of a group relying mainly on the division of responsibility to meet the
deadlines and to utilize available resources efficiently. I also learned how different
techniques were judged in relation to our final end requirement and then the
ideal one was chosen based on how each of them fared in the comparison.