## Inverse-RIAA filter

This is an RIAA "stripper" filter. Alternatively, you can think of it as an RIAA *encoder*.
Such a process has various applications, the most obvious of which being to "strip" RIAA de-emphasis from a needle-drop recorded via an RIAA preamplifier. This is useful if RIAA de-emphasis is *not* the correct equalisation; for example, if the record was recorded with one of the other historic recording characteristics. But there are many other applications in (for example) mastering and audio testing.

**The RIAA strip is especially useful if you already have a high-quality needle-drop solution. In this case, you can use the Inverse-RIAA filter to "strip" RIAA de-emphasis in order to apply the correct replay characteristic**

In the case of using the Inverse-RIAA filter to "strip" RIAA pre-emphasis in order to apply an alternative replay characteristic, it is actually better to record the needle-drop via a preamplifier with a flat frequency-response and subsequently to apply the correct equalisation using **Stereo Lab**. But this may not always be possible, in which case the Inverse-RIAA filter is useful as a first-stage process; the resulting file being subsequently processed by another of the phonograph filters.

**The RIAA recording and replay characteristics**

### Implementation

Note that the Inverse-RIAA filter in **Stereo Lab** is implemented *exactly as if it was an analogue pre-emphasis network*. It therefore introduces a complementary *phase response* as well as a complementary frequency response (see illustrations below). So the Inverse-RIAA filter entirely and cleanly "strips" the effect of the RIAA replay equalisation.

**The effect of RIAA encoding is non-phase linear (the effect upon a **^{(sin x)}/_{x} pulse is illustrated, left). The effect of decoding (also non phase-linear) "reverses" the phase distortion and restores an end-to-end, linear phase-response (right).

### Accuracy

Ideally the RIAA recording characteristic specification requires that the HF boost "goes on for ever"! The transfer function of the equalisation on the record side is given by,
**
H(jω) = [(1 + jω***T*_{1}) (1 + jω*T*_{3})] / (1 + jω*T*_{2})

where *T*_{1} = 3180µS, *T*_{2} = 318µS , and *T*_{3} = 75µS

From which it's fairly easy to see that **H(jω) ⇒ ∞ as ω ⇒ ∞. **

This isn't even possible in analogue circuitry (see here), and it certainly isn't possible in a sampled system. There is thus a necessity to adjust the transfer-function of the recording characteristic in the top three octaves (2kHz to 20kHz).

Nevertheless, the accuracy of an RIAA encode-decode performed at 48kHz sampling is measures better than ±0.15dB: several times better than we would normally expect of an analogue implementation.

**
Sweep test of RIAA encode-decode accuracy
**

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