THEORETICAL DISCUSSION

Coding converts the analog signal to be transmitted, generally a voice signal

coming from the telephone set through the SLIC TXA output, into serial digital
codes that are sent to the digital switching circuit of the central office via the
transmit (TX) terminal of the line interface. This process is generally known as
analog-todigital (A/D) conversion, and is referred to as voice digitization in the
telephone industry. Coding also converts serial digital codes received from the
central office circuitry through the receive terminal (RX) of the line interface, into
an analog signal that is sent to the telephone set via the SLIC RXA input. This
process is generally known as digital-to-analog (D/A) conversion, and is referred
to as voice recovery in the telephone industry.
Pulse-code modulation (PCM) is a method used to digitally represent
sampled analog signals. It is the standard form of digital audio in
computers, compact discs, digital telephony and other digital audio applications. In
a PCM stream, the amplitude of the analog signal is sampled regularly at uniform
intervals, and each sample is quantized to the nearest value within a range of
digital steps.
Starting with the introduction of the transistor, invented in 1947 by Bell
Laboratories, to amplification and switching circuits in the 1950s, and through
development of computer-based electronic switching systems, the public switched
telephone network (PSTN) has gradually evolved towards automation and
digitization of signaling and audio transmissions.
Digital telephony is the use of digital electronics in the operation and
provisioning of telephony systems and services. Since the 1960s a digital core
network has replaced the traditional analog transmission and signaling systems,
and much of the access network has also been digitized.
Digital telephony has dramatically improved the capacity, quality, and cost
of the network. End-to-end analog telephone networks were first modified in the
early 1960s by upgrading transmission networks with Digital Signal 1 (DS1/T1)
carrier systems, designed to support the basic 3 kHz voice channel by sampling the
bandwidth-limited analog voice signal and encoding using PCM. While
digitization allows wideband voice on the same channel, the improved quality of a
wider analog voice channel did not find a large market in the PSTN.
DIAGRAM/PICTURES/GRAPH/
CHARTS/TABLES

PROCEDURE 8

PROCEDURE 13: POSITIVE AND NEGATIVE VOLTAGE

PROCEDURE 15 PROCEDURE 17
EXPERIMENTAL DISCUSSION
In this experiment, we observed how the CODEC converts an analog signal
into a PCM Code. In the first part of the experiment, we set up a central office with
the Telephony Training System (TTS).
Then, we proceeded to voice digitization. We made sure that the address of
the TSAC in ALI A is set to 01 and that the two telephones can establish a
connection. We turned off the companding in the CODECs of the ALI A and B.
Then we connected the oscilloscope probes 1, 2 and 3 to the CODEC analog input,
CODEC digital output and TP17 respectively.
After setting the specific parameters, we set the output voltage of the DC
SOURCE of the ALI A to -0.5 volts. We observed in the oscilloscope screen that
the digital output of the CODEC is the DC SOURCE output voltage. Then we
measured the sampling interval and frequency, 125.33 microseconds and 8 kHz.
We varied the DC SOURCE output voltage slightly and we observed that
the CODEC digital output is the binary form of the output voltage. We also
observed that the most significant bit of the PCM Code determines the polarity of
the voltage converted. The arrangement of the least significant bits changes when
the DC SOURCE output voltage is varied. We observed that the voltage at the
CODEC analog input is directly proportional to the value of the serial PCM code.
In the last part of the experiment, we disconnected the CODEC analog input
from the DC source, and reconnected this input to the SLIC TXA output. Then we
spoke in the telephone handset while observing the original and recovered signals
as well as the serial PCM codes.
CONCLUSION
After performing the experiment, I conclude that the analog voice signals are
converted into serial digital codes with the use of pulse code modulation before the
signals are sent to the digital switching circuit of the central office, this is called
voice digitization. The digital voice signals are converted into analog signals from
the digital switching circuit to the analog telephone line, this is called voice
recovery. The pulse code modulation is performed by the CODEC in the analog
line interface. I also conclude that the CODEC input voltage is directly
proportional to the values of the serial PCM codes. The most significant bit of the
PCM code determines the polarity of the signal input at the CODEC.
GLOSSARY
Bit - is a basic unit of information used in computing and digital communications
Central Office (CO) - is an office in a locality to which subscriber home and
business lines are connected on what is called a local loop. The central office has
switching equipment that can switch calls locally or to long-distance carrier phone
offices.
CODEC - a device or program that compresses data to enable faster transmission
and decompresses received data.
Digitization - is the process of converting information into a digital (i.e. computer-
readable) format, in which the information is organized into bits.
Oscilloscope - is a laboratory instrument commonly used to display and analyze
the waveform of electronic signals.
Pulse-code modulation (PCM) - is a method used to digitally represent
sampled analog signals
Sampling - is the reduction of a continuous-time signal to a discrete-time signal.
Telephone - is a telecommunications device that permits two or more users to
conduct a conversation when they are too far apart to be heard directly. A
telephone converts sound, typically and most efficiently the human voice, into
electronic signals that are transmitted via cables and other communication channels
to another telephone which reproduces the sound to the receiving user.
Telephone Line - is a single-user circuit on a telephone communication system.
This is the physical wire or other signaling medium connecting the user's telephone
apparatus to the telecommunications network, and usually also implies a
single telephone number for billing purposes reserved for that user.
Telephony - is the field of technology involving the development, application, and
deployment of telecommunication services for the purpose of electronic
transmission of voice, fax, or data, between distant parties.
REFERENCES
https://en.wikipedia.org/wiki/Pulse-code_modulation
https://en.wikipedia.org/wiki/Telephony#Digital_telephony
https://www.elprocus.com/pulse-code-modulation-and-demodulation/
https://www.quora.com/What-is-the-audio-codec-for-PCM
http://www.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/asterisk-
CHP-7-SECT-2.html
REVIEW QUESTIONS
1. What does voice digitization consist of?
Voice digitization consists of sampling the voltage of the analog voice signal
to be transmitted at regular intervals, associating each sample with a specific
voltage interval called quantization interval, and producing the digital code
corresponding to that interval.
2. What are the functions of the most significant bit (MSB) and the other bits
in the serial PCM codes produced by a CODEC?
MSB is the bit indicating the polarity of the sample (sign bit), and the seven
other bits encoding the magnitude of the sample
3. What does voice recovery consist of?
Voice recovery consists of decoding each of the received digital codes to
find the corresponding quantization interval, and producing a voltage
corresponding to that interval.
4. What is the sampling frequency in the PCM CODEC used in analog line
interfaces?
In telephone systems, sampling is performed at a frequency (fs ) of 8 kHz.
5. Describe the bit format of the serial PCM codes produced by the CODEC
used in analog line interfaces.
Eight bits in each parallel PCM code are output one bit at a time, starting
with the MSB. In other words, the bit indicating the polarity of the sample (sign
bit) is output first, and the seven other bits encoding the magnitude of the sample
are then output successively.