Description
Background
Sounds, such as speech and music, are signals that change over time. The amplitude of a signal
determines the volume at which we hear it. The way the signal changes over time determines the type
of sounds we hear. For example, an ’ah’ sound is represented by a waveform shown in Figure 1.
Figure 1. A waveform for an ‘ah’ sound
The waveform is an analog signal, which can be stored in a digital form by using a relatively small
number of samples that represent the analog values at certain points in time. The process of producing
such digital signals is called sampling.
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Figure 2. A sampled waveform for an ’ah’ sound.
The points in Figure 2 provide a sampled waveform. All points are spaced equally in time and they
trace the original waveform.
The DE1-SoC board is equipped with an audio CODEC capable of sampling sound from a
microphone and providing it as input to a circuit. By default, the CODEC provides 48000 samples per
second, which is sufficient to accurately represent audible sounds.
This lab involves the design of several circuits that take input from a microphone through the CODEC,
record and process this sound data, and then play it back through speakers. To simplify the task, a
simple system that can record and playback sounds on the board is provided as a “starter kit”. The
system, shown in Figure 3, comprises a Clock Generator, an Audio CODEC Interface, and an
Audio/Video Configuration modules.
Figure 3. Audio System for this lab
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The left-hand side of Figure 3 shows the inputs and outputs of the system. These I/O ports supply the
clock inputs, as well as connect the Audio CODEC and Audio/Video Configuration modules to the
corresponding peripheral devices on the DE1-SoC board. In the middle of the figure, a set of signals to
and from the Audio CODEC Interface module is shown. These signals allow the circuit depicted on
the right-hand side to record sounds from a microphone and play them back via speakers.
The system works as follows. Upon reset, the Audio/Video Configuration begins an autoinitialization
sequence. The sequence sets up the audio device to sample microphone input at a rate of 48kHz and
produce output through the speakers at the same rate. Once the autoinitialization is complete, the
Audio CODEC begins reading the data from the microphone once every 48000th of a second, and
sends it to the Audio CODEC Interface core in the system. Once received, the sample is stored in a
128-element buffer in the Audio CODEC Interface core. The first element of the buffer is always
visible on the readdata_left and readdata_right outputs when the read_ready signal is asserted. The
next element can be read by asserting the read signal, which ejects the current sample and a new one
appears one or more clock cycles later, if the read_ready signal is asserted.
To output sound through the speakers a similar procedure is followed. Your circuit should observe the
write_ready signal, and if asserted write a sample to the Audio CODEC by providing it at the
writedata_left and writedata_right inputs and asserting the write signal. This operation stores a sample
in a buffer inside of the Audio CODEC Interface, which will then send the sample to the speakers at
the right time.
A starter kit that contains this design is provided as part of this lab.
Figure 4. Connections between FPGA and Audio CODEC
Task 1
You are to make a simple modification to the provided starter kit circuit to pass the input from the
microphone to the speakers. You should take care to read data from and write data to the Audio
CODEC Interface only when its ready signals are asserted.
Compile your circuit and download it onto the DE1-SoC board. Connect microphone and speakers to
the Mic In (pink) and Line Out (green) ports of the board and speak to the microphone to hear your
voice through the speakers.
Note: you may hear a lot of noise, or your speaker might be very quiet. Try listening in a quiet place
and/or using some 3.5mm headphones.
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Task 2
In this task, you will learn a basic signal processing technique known as filtering. Filtering is a process
of adjusting a signal – for example, removing noise. Noise in a sound waveform is represented by
small, but frequent changes to the amplitude of the signal. A simple logic circuit that achieves the task
of noise-filtering is an averaging Finite Impulse Response (FIR) filter. The schematic diagram of the
filter is shown in Figure 5.
Figure 5. A simple averaging FIR filter
An averaging filter, like the one shown in Figure 55, removes noise from a sound by averaging the
values of adjacent samples. In this particular case, it removes small deviations in sound by looking at
changes in the adjacent 8 samples. When using low-quality microphones, this filter should remove the
noise produced when you speak to the microphone, making your voice sound clearer.
You are to implement the circuit shown in Figure 5 to process the sound from the microphone, and
output the filtered sound through the speakers. Do you notice any difference between the quality of
sound in this part as compared to Task 1? (Provide your answer in the lab report).
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Figure 6. Circuit to generate some noise.
NOTE:
It is possible to obtain high-quality microphones with noise-canceling capabilities. In such
circumstances, you are unlikely to hear any effect from using this filter. If this is the case, we suggest
introducing some noise into the sound by adding the output of the circuit in Figure 6 to the sample
produced by the Audio CODEC.
module noise_generator (clk, enable, Q);
input logic clk, enable;
output logic [23:0] Q;
logic [2:0] counter;
always_ff @(posedge clk)
if (enable)
counter = counter + 1’b1;
assign Q = {{10{counter[2]}}, counter, 11’d0};
endmodule
The circuit is a simple counter, whose value should be interpreted as a signed value. The circuit should
be clocked by a 50MHz clock, and the enable signal should be driven high when the Audio CODEC
module can both produce and accept a new sample.
To hear the effect of the noise generator, add the values produced by the circuit to each sample of
sound from the Audio CODEC in the circuit in task 1.
Task 3
The implementation of the averaging filter in task 2 may have been effective in removing some of the
noise, and all of the noise produced by the noise generator. However, if your microphone is of lowquality or you increase the width of the counter in the noise generator, the filter in task 2 would be
insufficient to remove the noise. The reason for this is because the filter in task 2 only looked at a very
small time frame over which the sound waveform was changing. This can be remedied by making the
filter larger, taking an average of more samples.
In this task, you are to experiment with the size of the filter to determine the number of samples over
which you have to average sound input to remove background noise. To do this more effectively, use
the design of an averaging FIR filter shown in Figure Error! Reference source not found..
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Figure 7. N-sample averaging FIR filter.
To compute the average of the last N samples, this circuit first divides the input sample by N. Then,
the resulting value is stored in a First-In First-out (FIFO) buffer of length N and added to the
accumulator. To make sure the value in the accumulator is the average of the last N samples, the
circuit subtracts the value that comes out of the FIFO, which represents the (n+1)
th sample.
Implement, compile and download the circuit the DE1-SoC board. Connect microphone and speakers
to the Mic and Line Out ports of the board and speak to the microphone to hear your voice through the
speakers. Experiment with different values of N to see what happens to your voice and any
background noise, remembering to divide the samples by appropriate value. We recommend
experimenting with values of N that are a power of 2, to make the division easier.
Use the SignalTap II functionality of Quartus to verify the contents of your FIFO buffer.
If you have a portable music player, with a connector such that you can supply input to your circuit
through the Mic port, try experimenting with different sizes of the filter and its effect on the song you
play. Provide the results of your experimentation in the lab report.
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Lab Demonstration and Submission Requirements
• Submit a lab report that includes the procedures and results obtained in the lab. Suggestions on what to
include follow:
o Abstract: A brief overview of the entire report
o Introduction: What the lab is (specifications or background info, etc.
o Procedure/Results/Analysis:
▪ Steps to complete the lab
▪ Description of each module (explain how they work, point out code, etc.)
▪ Simulations of each module with comments on important things to notice about them.
▪ Overall description of what the finished product was/how it behaved
▪ Discussions of any problems had while completing the lab, or unsolved errors
o Conclusion: A final summary, reflection on the lab or what was learned, etc.
• Include any hurdles or challenges (if any) that you faced in finishing this lab and how you overcome
them.
• Submit the SystemVerilog code for all tasks and include screenshots for the ModelSim
waveforms of your modules and your signal tap results.
• Submit the Flow Summary (produced during compilation) of compiling your system.
• In your report, include the number of hours (estimated) it took to complete this lab, including
reading, planning, design, coding, debugging, testing, etc. Everything related to the lab (in total).
• Submit your report and programs to Canvas. No hard copies. Submit a pdf file as well as a
compressed folder of your source files.
