Fullscreen HTML5Med Res (??MB)HTML4

AUDIO OUT AUDIO OUT L R By Jake Rothman Analogue Vocoder – Part 1 T ime for a new music project! Many of you are probably familiar with the vocal effect of an electronic instrument called the ‘vocoder’. It’s a fascinating piece of electronics, and over the next few months we will design and build a very high-quality example. The vocoder’s primary claim to fame is where human speech is superimposed on a musical instrument. A popular example is the creepy 1961 recording Sparky’s Magic Piano – see https://youtu.be/ Km19Iohd1YA (strictly speaking, Sparky’s Piano used a Sonovox which was not a vocoder, but a pair of special loudspeakers (compression drivers) applied to the vocalist’s throat. It was also used in Dumbo for the talking train whistle.) A more recent example is the definitive vocoder song, Laurie Anderson’s 1981 release, O Superman – https://youtu.be/ S39NaDPNDtk (see our editor’s prized, pristine 12-inch vinyl copy in Fig.1!). It was recorded with the Roland VP-330: www. vintagesynth.com/roland/vocoder.php Origins As with most electronic audio/music technology, vocoders were originally developed for telecommunications. Invented by Homer Dudley at Bell Labs in Fig.1. Laurie Anderson’s O Superman, a vocoded treat in avant guard minimalism using the Roland VP-330. Analysi s f ilters ( double- tuned) 1936, they enabled more speech channels to be put through long low-bandwidth cables and other transmission systems. Vocoders are an analogue data compression system which operate by representing speech data in its simplest form, amplitude changes of frequency bands, akin to a spectrum analyser. These slow changes can then be transmitted and the speech resynthesised at the other end. The military used them to disguise voices and encode secret messages. This is where the name comes from: VOice enCODER. A vocoder can be thought of as a system for transferring the spectral energy of a speech signal onto another sound. An unfiltered organ or string synthesiser works especially well. If white noise is used, the vocoded voice will sound like a whisper. An analogue vocoder is complex, the design I’m presenting consists of 28 precision filters and 14 voltage-controlled amplifiers (VCAs). For a ‘normal’ Smoothing R ectif ier low- pass f ilters f1 D rive amplif ier Multiple channels C ontrol vo ltages to V C A Speech input ( modulation) f1 2 I N P U T SE C T I O N O U T P U T SE C T I O N Analysi s f ilters ( double- tuned) f1 V C A D rive amplif ier Mixe r Multiple channels Fig.2. Software-based vocoders are possibly the cheapest entry point for most musicians, assuming you have a powerful computer and digital audio workstation software installed. Ironically, they can be a good way of optimising an analogue vocoder design before building. 48 C arrier ( string/ syn th/ paino) f1 2 + V ocoder output V C A Fig.3. Analogue Vocoder block diagram, showing analysis, control and resynthesis sections. Practical Electronics | November | 2021 n V ocoder channel card Speech fL I N P U T SE C T I O N f L and f U are the lower/ upper f req uencies of the double- tunded band- pass f ilter fU V - to- I conve rter O U T P U T SE C T I O N Syn th fL V C A fU O utput D ual V C A shared with another channel L R System architecture Mixe r busses selectable f or lef t, right or both Fig.4. Each vocoder channel consists of these internal building blocks. electronic engineer this is obviously something best achieved by digital signal processing using FFT (Fast Fourier Transform) and DSP hardware and software. Vocoder software plug-ins are the entry point for most musicians wanting a vocoder sound. In my case, I played about with the digital Vocodex Channel Vocoder in FL Studio (Fig.2) to find the best frequencies and Q values to use. However, digital vocoders have their own problems. They need a lot of computing power and often exhibit considerable time delay or latency. They all use similar maths which imparts a peculiar ‘under water’ colouration at low signal levels, which I find unpleasant. Most smartphones use vocoding as part of their compression algorithms, so most people will be familiar with this nasty effect. Software vocoders brought out a £400 Chinese surface-mount copy of the Roland, called the VC340. Still, my design is stereo and it’s modifiable – important features, because having one’s own unique sound signature is an essential attribute for electronic musicians. are also complex to use with too many features and parameters to be set up. On the other hand, analogue vocoders have an immediate playability that lends them to composition and live performance. DIY analogue vocoders are expensive to build, typically costing a few hundred pounds. However, buying a ready-made one, such as the Roland VP-330 Vocoder Plus, or Tim Orr’s Electronic Music Studios (EMS), will set you back several thousand pounds: https://bit.ly/pe-nov21-voc We are getting to the point where analogue electronic musical instruments are becoming expensive antiquities in their own right. This is one of the few occasions in electronics where building ones own can save serious money. Ironically, just as we go to press, and after spending five years building this Vocoder, Behringer have The input part of a vocoder is concerned with analysing the changes of a signal, usually speech in musical applications, and often called the ‘modulator’ in communications. This section generates slowly changing DC control voltages representing the amplitudes of the different frequency bands. The output part is concerned with the generation or synthesis of the final sound. The block diagram is shown in Fig.3. In music applications, chords, from say a piano or string synthesiser, (or the carrier/ excitation in comms) are split into corresponding frequency bands. These bands are then modulated by the control signals derived from the input section. This requires a VCA to control the output of each frequency band. The end result is that the piano or other musical input can be made to ‘talk in tune’. Alternatively, the control signals can be patched from one frequency band to another to make crazy noises and facilitate speech scrambling. The internals of the frequency-band modules are shown in the block diagram Fig.5. Early photo of the prototype analogue vocoder. No knobs yet. Practical Electronics | November | 2021 49 f L ( – 3dB) = 4 4 6 H z fL = 4 7 2 H z f U = 52 9 H z f U ( – 3dB) = 56 0 H z f c= 50 0 H z 0 dB = Bandwidth = 1 / 3 octave ( typ ical f or vo coder) Single Q = 4 . 38 f ilter at 50 0 H z fL – 3dB Q fU = f c / ( f U ( – 3dB) – f L ( – 3dB) ) = 50 0 / ( 56 0 – 4 4 6 ) = 4 . 38 – 3dB Fig.6. Prototype Vocoder channel PCB. Each channel frequency has to have different values of filter capacitors. (The final version will be double-sided to remove all those links.) and are further expanded in Fig.4. Each of these are made into a PCB card that can be plugged into a bus-board to connect them all together. The prototype Analogue Vocoder is shown in Fig.5. This design uses 12 band-pass filter modules with frequencies optimised for speech; plus, the system topped and tailed by fourth-order low-pass (fc = 120Hz) and high-pass filters (fc = 8kHz). This builds to a total of 14 filters, all fourth-order, and requiring 112 close-tolerance capacitors which represent a major chunk of the parts bill. The EMS vocoder has 22 band-pass filters! A prototype band-pass vocoder channel board is shown in Fig.6. Band-pass filters There are many ways to build a band-pass filter. The state-variable and its alternative the bi-quad are very effective. The bi-quad’s resonant frequency (fc) can be adjusted with just a single resistor; and the state-variable’s fc with two resistors – see Fig.7. However, both these configurations need three op amps per filter, which would be excessive in a multiple filter system like a vocoder. For perfect frequency band discrimination, digital brick-wall band-pass filters are used – but I suspect in musical applications these would sound horrible because of the group-delay-induced ringing. Ω Ω Y Better bring in Q, Bond The band-pass frequencies in the speech band need to be spaced at around one third of an octave steps. This necessitates a filter Q of around 4.38, to put the cutoff (−3dB) frequencies in the right place. To obtain a flatter overall frequency response, double-tuned band-pass filters can be used, which comprise two filters whose curves overlap, as shown in Fig.8. These have two resonances close together with a Q of 8.8, which gives a ‘flat’ top with initial steep (−40dB per octave) 4 . 3nF R 2 R – + R Fig.8. By putting two high Q band-pass filters in series a good compromise curve can be obtained with steep skirts and a flat top. This forms a double-tuned bandpass filter, familiar to most radio designers. Ω – I nput R – + R Bandpass output 1 1 = R R – + O ptional input attenuator resistor f 1 √R 1 R 2 = Y 1 R – Ω – + + 1 1 R 2 2 R 1 2 2 Ω I nput O utput 0 V X R Ω 4 . 3nF Ω + Ω X I nput R 1 Ω f Ω f L and f U are the lower/ upper f req uencies of the double- tuned band- pass f ilter f Altering this resistor changes f c R 1 = – a lot! ) 1 √R 1 R 2 = f 1 = 4 . 4 2 nF + Fig.7. The state-variable (above) and bi-quad (below) are the ideal building blocks for band-pass filters but are too complex and expensive for the multiple channels needed in a vocoder. However, for a vocoder with a few channels where the parameters need front panel controls, the bi-quad would be used. Fig.9. The multiple-feedback filter, the simplest and cheapest. High gain makes it noisy. Two of these are needed to make the flat-topped filter. An input attenuation resistor to ground is often added, but the parallel resistance of the attenuator must still equal 3.6kΩ to avoid altering the response. 50 Practical Electronics | November | 2021 f = / R Bandpass output Q 86 0 H z, = 1 0 Q R 3 I nput R 1 C 1 33nF Ω R 2 C 4 33nF Ω R 6 R 4 – + Ω R 5 = 1 0 C 2 33nF Ω – R 1 + Ω Fig.10. A double-tuned band-pass filter circuit. One-third octave bandwidth with a centre frequency of 950Hz. 8kH C f I nput 2 . 2 nF R f C f 2 . 2 nF z high- pass f ilter Ω Ω 2 . 2 nF 2 . 2 nF + IC 1 a – R f F or speech analysi s f ilters use T L 0 82 , and f or the output section use low- noise N E 5532 devi ces. O utput IC 1 b Ω – Ω O p amp selection Ω Ω Ω 2 7 0 pF Ω 2 7 0 pF 0 V 1 2 0 H z low- pass f ilter R f 1 µ F I nput Ω Ω R f Ω C f 2 2 0 nF C f 2 2 0 nF 2 2 0 nF + Ω Ω IC 2 b – + IC 2 b 2 2 0 nF Ω – Ω Ω Ω Ω O utput C 2 1 0 0 nF 0 V = 1 fC C L et f C = 1 kH z Ω = 1 0 0 nF C P eak gain = 2 f or this f ilter Ω C 1 - C 2 = 1 0 0 nF Ω F or a typ ical vo coder Q of 5: Ω Ω Fig.11. The dual-amplifier band-pass configuration filter. Twice as many op amps, but more controlled gain. dual-amplifier band-pass configuration (DABP) developed by Sedra and Espinoza, and shown in Fig.11. This avoids the excessive noise gain problem of the multiple-feedback type. However, it adds 24 extra op amp sections to the vocoder, so we’ll leave this for a future upgrade. High / low-pass filters The high-pass and low-pass filter topology is basically the same as the vocoder channel topology (Fig.4), but with the band-pass filters replaced with fourth-order high-pass and low-pass filters. These filter circuits are shown in Fig.12. These extra modules are only needed if a full frequency response is desired for a stand-alone-performance vocoder. In a band or studio situation where there is a bass player and drummer for example, the vocoder only needs to cover the speech band – ie, the middle frequencies of around 100Hz to 8kHz. A treble-boosted portion of direct speech signal can be mixed in to give O utput a clean top-end. By eliminating the high-pass and low-pass vocoder modules one saves £70, and gains a less cluttered mix into the bargain. 0 V N ote: same P C B can be used f or both circuits, j ust swap R f and C f positions to make low- pass or high- pass f ilter card Fig.12. For the top and bottom bands of a full-range vocoder, fourth-order low-pass and high-pass filters are used. These require a different PCB and are not needed for a speech-only vocoder. Practical Electronics | November | 2021 R 3 + R simple design is that the gain is 50 at a Q of 5, which is a problem because the op amp output will clip. For third-octave double-tuned filters, the Q of each section needs to be 8.8, the practical limit for this type of filter. A circuit of a double-tuned multiple feedback filter is shown in Fig.10. The signal has to be attenuated on the input to the bandpass filters by even more than for a single filter. Therefore, the signal-to-noise ratio is poor. This has little effect in speech filters, since these are followed by full-wave rectifiers to produce smoothed DC control voltages for the VCAs. On the carrier/output section it is more significant, but it is only the output filters after the VCAs that contribute continuous noise. The noise from the filter stages preceding the VCAs is effectively muted when the VCAs cut off. An alternative filter, which I intend to try at some point in the future, is the + – Ω D C path to ground req uired 0 V The circuit shown in Fig.9 is the simplest band-pass filter; it’s called the multiple-feedback filter because it uses two feedback paths. The disadvantage of this R 2 Ω I nput C entre f req uency of whole ( f lat- topped) f ilter is 9 50 H z, total Q of f ilter is 5. Multiple-feedback band-pass filter 0 V O utput 0 V skirts. The result is much less peaky than a single filter and the out-of-band attenuation is double. The frequencies of the two filters are selected so as to correspond with what would be the −3dB points of a single filter. So for a 500Hz (centre frequency) double-tuned filter, the resonant frequencies (fL and fU) of the two filters would be 472HZ and 429Hz respectively. Note the slight asymmetry is due to the neeed to match (approximately) exponential frequency scaling. The double-tuned technique was originally developed for radio intermediate frequency transformers in superhet radios. C 1 1 0 0 nF Ω C 3 33nF Ω R 5 Ω R 4 – z, + 1 . 2 5kH Next month That’s all for this month – in the next issue we’ll start to look in more depth at the thinking behind the design. 51