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Analogue to Digital Conversion

By , Thursday 17th October 2002 in Hardware

In converting an analogue signal to a digital format will require first taking samples at regular intervals. At least two samples must be taken during each cycle of the analogue signal.

The sample level must be held in order that a comparison can be made against a range of test voltages, each of which corresponds to a unique digital code. Unless we use an infinite number of codes it will not usually be possible to obtain an exact match between the analogue sample level and the digital code level.

However, by using the nearest available level above and/or below will result in only a small imperfection known as quantisation error. In this way, the regular analogue signal samples are allocated a digital code, which can now be transmitted over the digital network. In the same way that computer generated data bytes are converted from parallel to serial format, so the quantised analogue samples are converted into a serial form to pass over a single network channel.

It may be necessary to add synchronising bits, error checking and/or correcting bits prior to transmission, but from this point onwards there is no essential difference between this converted analogue signal and a computer generated data signal.

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

If the analogue signal is between +5 volts and —5 volts and we are using an eight bit code with no redundancy, then the 256 available codes gives:- 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 be entering a digital system having a much wider bandwidth, where data rates in the order of Mbps are to be expected. This telephone service will be multiplexed along with other speech, video and fax services plus digital data traffic. It may well be compressed in a similar way to computer files in order to maximise the available bandwidth.

Other analogue speech circuits could be electronically switched to the input of the Analogue to Digital converter, which would process these in a similar manner. Because each would need to be sampled at the same rate (twice per cycle), then 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 along with 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 in order that it can be recovered correctly. This would result in some channels being unused at any given time and this would be an uneconomic use of available bandwidth.

An alternative method may be to have time slots for general use. Any of the services, phone, fax, video, data etc. will then 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 it requires additional information to 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 such 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 then all of 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 different services in the most efficient way to make best use of the bandwidth available.

Although data, such as a computer file, can be transferred at just about any rate, or even 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 of this when an interviewer uses a satellite link question a guest in some distant location. Due to the very long route the signal must take, there is a significant delay imposed between the two participants, which frequently cause problems. Interactions seem to be coming at inappropriate times and the usual smooth flow breaks down. If this is a problem for people who are experienced in such situations it would be catastrophic for the general public. For this reason, the way that phone circuits and data circuits are handled must have different considerations as they pass through the communication network. The data from a computer file can be separated into discrete blocks, follow different routes having different delays, be temporarily stored on route, before being finally received and reassembled in the correct order for use.

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

Although telephone speech signals have been referred to, these 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/or coding bits. Examples will include music, video and control signals.

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Tim Trott

Tim is a professional software engineer, designer, photographer and astronomer from the United Kingdom. You can follow him on Twitter to get the latest updates.

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