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www.americanradiohistory.comDSP THEFUTURE OFAUDIO CREATIVITYFrancis Rumsey discusses thedevelopment of digital signal processingand its implications for the futureDigital Signal Processing (DSP), iswithout a doubt the technology thatwill take the audio industry into thethe next century. It has implicationsas important as any of the major audiodevelopments of the last century in that it willchange the face of audio production and provideusers with a flexible tool for the modification ofsound fields. DSP relies on the processing of vastamounts of data at high speeds and thus isdirectly related to advances in computertechnology which the audio industry is able toencompass once sound is in digital form.An insight will be given into some of thecapabilities of DSP with an example taken fromthe interesting new AKG Creative AudioProcessor (CAP) in order to gain some ideas aboutthe direction in which we are heading.PrinciplesDSP is based on the mathematical manipulationof digital audio data. If a sequence of numbersrepresents a particular sound waveform, then bychanging the numbers one may change the sound.Clearly it is exactly how the numbers arechanged that dictates the audible result.Essentially, DSP relies on the principle ofdigital filtering, which involves the successivedelay of samples, multiplication by a co- efficient,then adding the result back into the data streamat a suitable point. Depending on the delay andco- efficients, equalisation networks can be builtup that simulate boost or cut at certainfrequencies; furthermore, the Q of the filter or theslope of an EQ curve depends on how many stagesof delay and multiplication are used. It isnecessary to realise that time delay without lossof sound quality is easy in the digital domain: itsimply means storing a set of audio samples for atime before they are read out of a memory, andthe only limit on the delay is the size of thememory. The advance of audio in time is arelative matter as clearly, true advance involvesan element of prophecy. It can also be simulatedby delaying everything except the signal that isto appear advanced, the only side -effect being anoverall delay between the sound entering theprocessor and that leaving it.78 Studio Sound, June 1989Since time delay is required for equalisation itis perhaps reasonable to assume that anyprocessor capable of EQ will be capable of othereffects as well, and this is indeed true. Many ofthe effects that audio engineers are familiar with,such as reverb, echo, flanging and so forth, areachieved through the use of delay, repetition, andlevel control of digital audio samples. To give asimple example, a digital echo effect could beachieved by reading samples from the output ofthe processor into a memory buffer and readingthem out again at the specified echo delay later;meanwhile multiplying the delayed sample valuesby an attenuating value (to simulate thereduction in level of the echo), then adding thedelayed signal back into the input. The delayedsignal plus the original signal would then appearat the output and the total would go round thedelay loop again, the original echo being echoedat a still lower level, at the same time as beingtapped off to the audio output.Time -related functions also come into pitch orfrequency control of signals, as it is possible tochange the pitch of a signal by altering the rateat which digital samples are converted. Unlessaction is taken this will also result in a change inthe apparent speed ofplayback, togetherwith the problem thatyou can't read samplesout of a processorfaster than the speedyou're writing them in.The action required isusually referred to as'sample rateconversion', andinvolves digitalfiltering principlessimilar to thoseindicated above, inorder to calculate thecorrect amplitudes ofsamples at the newsampling rate. Thisprocess also involvesdelay whilecalculations take place.In addition to time -related level control ofsignals we also have to consider level -related levelcontrol. In varying the multiplication co- efficientof samples depending on their level as opposed totheir time position it is possible to achieve effectssuch as compression and limiting. This is nothard to imagine, as the action of an ordinaryanalogue compressor is that of altering theamount of gain of an amplifier depending on thelevel of the audio signal. The correct combinationof time and level -related manipulation of audiosamples is the key to effective DSP.Digital mixersPeople talk a lot about digital mixers, askingwhen they will appear. From the above principlesit ought to be clear that the mixing of audiosignals is just another feature of DSP, and thatany processor capable of performing EQ andeffects will also be capable of level control andmixing: it only depends on the intention of thedesigner, and the software that is written. Oneshould not be surprised then that themanufacturers who are experimenting with `soft'DSP products (those with a graphic user interface)can show a small mixer screen, an effects screen,an equaliser screen and a pitch change screen(some of these may be combined).By way of explaining the digital mixing processone really only has to imagine again themultiplication of audio data for each channel by aco- efficient corresponding to the fader position,and the addition of time -coincident sample valuesfrom one channel to those from other channels inorder to mix them together. One side -effect ofadding lots of digital sample values together isthat the resulting binary numbers can get quitelarge; similarly, the result of digital EQ (beingthe adding back in of delayed and multipliedsample values to the original data stream) is alsothat larger numbers are produced. Although thewindows of any DSP system on the analogueworld (the converters) are limited to 16 or 18 bitaccuracy, internal data may have considerablylonger word -lengths than this because of theeffects of mathematical manipulation. This iswhat is meant when a manufacturer quotes 24 bitor 32 bit accuracy in a digital mixer. If soundquality is not to be impaired in the truncation ofinternal samples from this length to 16 bits forconversion, various intelligent roundingtechniques together with the use of digital ditherand oversampling may be used.An interesting observation which arises from aYamaha's DMP7 incorporates effects and automation
ill Audio lad.+iol0ngboume House,Hollingboume,{an_ MEI7 IQJ EnglandTelephone: E652 780) 55ETelex: 966641 H LLfax: (062 73C) 550.HILLAUDIOA rangc of six 2U Ampl fiers from 200w -nonc to 800.. per channel stereoFill Aucio la50028 N. Royal Atlanta Ca.Tucker,:34 30084 USA.Telephone: 404) 934 1851Telex: 2938:7 H_ADFzx:(404)534 1040.
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