20-151-0178 Rabbit Semiconductor, 20-151-0178 Datasheet - Page 68

CABLE CONVERTER RS-232 TO USB

20-151-0178

Manufacturer Part Number
20-151-0178
Description
CABLE CONVERTER RS-232 TO USB
Manufacturer
Rabbit Semiconductor
Datasheet

Specifications of 20-151-0178

Accessory Type
USB to RS232 Adapter
Product
Microcontroller Accessories
For Use With/related Products
Rabbit-based Boards
Lead Free Status / RoHS Status
Lead free / RoHS Compliant
Other names
20-151-0178
316-1181
5.6 Speaker
DAC Channel #0 can be configured as an input to a speaker. The audio hardware on the RabbitFLEX
board has two sections, the filter and the amplifier. The first part, the filter, converts the high speed digital
pulses that come from the Rabbit processor into a smoothly varying analog signal. It does this through low
pass filtering. According to its name, low pass filtering allows low frequency signals to pass through it
while blocking higher frequency signals. For a complex signal like digital pulses, this has the effect of
averaging them out to produce an analog signal. In this way, the low pass filter converts a pulse width
modulated (PWM) signal into a variable voltage signal. The filter is composed of resistors and capacitors
arranged so that the high frequency signals are bypassed to ground through the capacitors.
After the audio signal is filtered it must be amplified in order to be powerful enough to drive a speaker.
This is done with an LM386 audio amplifier IC. The signal from the filter must be amplified because it is a
high impedance signal. Simply put, a high impedance signal is one that is degraded if it is used to directly
power an electrical load. So instead of directly connecting the filter output to the speaker, it is connected to
the amplifier input. The amplifier input senses the signal from the output and generates a matching low
impedance signal that has enough power to drive the speaker effectively.
When dealing with digital to analog conversion and WAV files (a popular standard for digitized audio
information) one of the most important questions is does it sound like the original analog signal that was
used to create the WAV file?
The answer depends on two things: the sampling rate and the sampling precision used when creating the
digital audio file. Digital audio samples are simply a sequence of values, each value representing the
amplitude of an audio signal at a given point in time.
Figure 5.4 Sampling of a Sound Waveform
The sampling rate is the number of samples taken in one second. Looking at the figure above, you can see
that if you connect the dots from left to right with straight lines, you lose a lot of the information repre-
sented by the curved line. By increasing the sampling rate, the closer the straight lines match the curved
line, meaning less information is lost.
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