How Analogue to Digital Conversion Works

How Analogue to Digital Conversion (ADC) systems work to convert analogue signal into a digital format which can be transmitted on networks.

By Tim TrottHow Stuff Works • October 17, 2002
902 words, estimated reading time 3 minutes.
How Analogue to Digital Conversion Works

The crucial first step in converting an analogue signal to a digital format is regular sampling. This process necessitates taking at least two samples during each cycle of the analogue signal.

The sample level must be precisely held so that a comparison can be made against a range of test voltages, each corresponding to a unique digital code. Unless we use an infinite number of codes, obtaining an exact match between the analogue sample level and the digital code level will not usually be possible. This precision is crucial in ensuring the accuracy of the digital conversion process.

By using the nearest available level above and below, we can minimize the impact of quantisation error, which is a small imperfection. This allows the regular analogue signal samples to be allocated a digital code, ready for transmission over the digital network. Just as computer-generated data bytes are converted from parallel to serial format, the quantised analogue samples are transformed into a serial form to pass over a single network channel.

Before transmission, it's essential to incorporate synchronising bits, error checking, and correcting bits. These elements play a crucial role in maintaining the integrity of the converted analogue signal, making it indistinguishable from a computer-generated data signal.

The highest frequency in a telephone-quality speech signal is 3000 Hz. Hence, at least two samples for one cycle of this signal result in a sample rate of 6000 samples per second. This is to take a sample every 150ms.

If the analogue signal is between +5 and -5 volts and we use an eight-bit code with no redundancy, then the 256 available codes give:- 10/256 = 39 mV between levels.

These eight bits must be transmitted in the next 150 ms - so the data rate needs to be just over 50 kbps.

While this data rate is similar to that which is possible over the present telephone line, it is intended to enter a digital system with a much wider bandwidth, where data rates in the order of Mbps are expected. This telephone service will be multiplexed along with other speech, video and fax services, plus digital data traffic. It may be compressed similarly to computer files to maximise the available bandwidth.

Other analogue speech circuits could be electronically switched to the input of the Analogue to Digital converter, which would similarly process these. Because each would need to be sampled at the same rate (twice per cycle), each will generate eight (for example) data bits. Hence, if six telephone circuits are required, this will create 8 x 6 bits, which must be transmitted in 150 bits. The data rate is, therefore, 48 bits in 150ms or 320 kbps, which is still a modest rate for a digital system.

These would then be multiplexed into a common channel, and data from other services to form a data stream. Each service could have a specified time slot within this data stream, which must be known to the receiving terminal so that it can be recovered correctly. This would result in some channels being unused at any given time, which would be an uneconomic use of available bandwidth.

An alternative method may be to have time slots for general use. Any services, phone, fax, video, data, etc., will be allocated to any available time slot within the data stream until all are in use.

Although this makes better use of the available resources, additional information must be sent to identify what each time slot is being used for. This system also allows for "intelligent" control where, for example, all phone calls can be given priority access to the time slots so that no matter how busy the system is, all incoming and outgoing phone calls are not delayed. On the other hand, where a data link needs to be made, if several time slots are available, these can be used to create a high-speed circuit. The flexibility of this type of multiplexing allows circuits to be established for a range of services in the most efficient way to make the best use of the bandwidth available.

Although data, such as a computer file, can be transferred at any rate or paused, this does not apply to a phone circuit. There is an international agreement limiting the delay on a phone circuit since this can cause the service to become unacceptable. You will encounter an example when an interviewer uses a satellite link to question a guest in some distant location. Due to the very long route the signal must take, a significant delay is imposed between the two participants, which frequently causes problems. Interactions come at inappropriate times, and the usual smooth flow breaks down. If this is a problem for people experienced in such situations, it would be catastrophic for the general public. For this reason, how phone and data circuits are handled must be considered differently as they pass through the communication network. The data from a computer file can be separated into discrete blocks, followed by different routes with different delays, and temporarily stored on the route before being received and reassembled in the correct order.

This is impossible with the data from a phone circuit, which must remain in the correct sequence and not be subject to any significant delay.

While telephone speech signals have been the focus, it is important to note that they are not the only analogue signals that may need to be transmitted over the network. The method of digitising these will be the same but may require different sample rates and coding bits. This broad range of signals, including music, video, and control signals, underscores the complexity and depth of our topic.

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