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Commuted Piano Synthesis

The general technique of commuted synthesis was introduced in §8.7. This method is enabled by the linearity and time invariance of both the vibrating string and its acoustic resonator, allowing them to be interchanged, in principle, without altering the overall transfer function [449].

While the piano-hammer is a strongly nonlinear element (§9.3.2), it is nevertheless possible to synthesize a piano using commuted synthesis with high fidelity and low computational cost given the approximation that the string itself is linear and time-invariant9.4.2) [467,519,30]. The key observation is that the interaction between the hammer and string is essentially discrete (after deconvolution) at only one or a few time instants per hammer strike. The deconvolution needed is a function of the hammer-string collision velocity $ v_c$. As a result, the hammer-string interaction can be modeled as one or a few discrete impulses that are filtered in a $ v_c$-dependent way.

Figure 9.31 illustrates a typical series of interaction forces-pulses at the contact point between a piano-hammer and string. The vertical lines indicate the locations and amplitudes of three single-sample impulses passed through three single-pulse filters to produce the overlapping pulses shown.

Figure 9.31: Overlapping hammer-string interaction force pulses.

Force-Pulse Synthesis

Figure 9.32: Creating a single hammer-string interaction force-pulse as the impulse response of a filter. The filter depends on the hammer-string collision velocity $ v_c$, but it is LTI while $ v_c$ is fixed.

The creation of a single force-pulse for a given hammer-string collision velocity $ v_c$ (a specific ``dynamic level'') is shown in Fig.9.32. The filter input is an impulse, and the output is the desired hammer-string force pulse. As $ v_c$ increases, the output pulse increases in amplitude and decreases in width, which means the filter is nonlinear. In other words, the force pulse gets ``brighter'' as its amplitude (dynamic level) increases. In a real piano, this brightness increase is caused by the nonlinear felt-compression in the piano hammer. Recall from §9.3.2 that piano-hammer felt is typically modeled as a nonlinear spring described by $ f(x)=k\,x^p$, where $ x$ is felt compression. Here, the brightness is increased by shrinking the duration of the filter impulse response as $ v_c$ increases. The key property enabling commuted synthesis is that, when $ v_c$ is constant, the filter operates as a normal LTI filter. In this way, the entire piano has been ``linearized'' with respect to a given collision velocity $ v_c$.

Multiple Force-Pulse Synthesis

One method of generating multiple force pulses as shown in Fig.9.31 is to sum several single-pulse filters, as shown in Fig.9.33. Furthermore, the three input impulses can be generated using a single impulse into a tapped delay line2.5). The sum of the three filter outputs gives the desired superposition of three hammer-string force pulses. As the collision velocity $ v_c$ increases, the output pulses become taller and thinner, showing less overlap. The filters LPF1-LPF3 can be given $ v_c$ as side information, or the input impulse amplitudes can be set to $ v_c$, or the like.

Figure 9.33: Multiple hammer-string interaction force-pulse as the superposition of impulse-responses of a parallel lowpass-filter (LPF) bank.

Commuted Piano Synthesis Architecture

Figure 9.34 shows a complete piano synthesis system along the lines discussed. At a fixed dynamic level, we have the critical feature that the model is linear and time invariant. Therefore we may commute the ``Soundboard & Enclosure'' filter with not only the string, but with the hammer filter-bank as well. The result is shown in Fig.9.35.

Figure 9.34: Piano synthesis using natural ordering of all elements.

Figure 9.35: Piano synthesis using commuted ordering.

As in the case of commuted guitar synthesis (§8.7), we replace a high-order digital filter (at least thousands of poles and zeros [229]) with a simple excitation table containing that filter's impulse response. Thus, the large digital filter required to implement the soundboard, piano enclosure, and surrounding acoustic space, has been eliminated. At the same time, we still have explicit models for the hammer and string, so physical variations can be implemented, such as harder or softer hammers, more or less string stiffness, and so on.

Even some resonator modifications remain possible, however, such as changing the virtual mic positions (§2.2.7). However, if we now want to ``open the top'' of our virtual piano, we have to measure the impulse response that case separately and have a separate table for it, which is not a problem, but it means we're doing ``sampling'' instead of ``modeling'' of the various resonators.

An approximation made in the commuted technique is that detailed reflections generated on the string during hammer-string contact are neglected; that is, the hammer force-pulse depends only on the hammer-string velocity at the time of their initial contact, and the string velocity is treated as remaining constant throughout the contact duration.

Progressive Filtering

To save computation in the filtering, we can make use of the observation that, under the assumption of a string initially at rest, each interaction pulse is smoother than the one before it. That suggests applying the force-pulse filtering progressively, as was done with Leslie cabinet reflections in §5.7.6. In other words, the second force-pulse is generated as a filtering of the first force-pulse. This arrangement is shown in Fig.9.36.

Figure 9.36: Commuted piano synthesis supporting three hammer-string interaction pulses using separate filters for each pulse and implementing the filters successively. Each new delay is equal to the travel from the hammer, to the agraffe, and back to the hammer.

With progressive filtering, each filter need only supply the mild smoothing (and perhaps dispersion) associated with traveling from the hammer to the agraffe and back, plus the mild attenuation associated with reflection from the felt-covered hammer (a nonlinear mass-spring system as described in §9.3.2).

Referring to Fig.9.36, The first filter LPF1 can shape a velocity-independent excitation signal to obtain the appropriate ``shock spectrum'' for that hammer velocity. Alternatively, the Excitation Table itself can be varied with velocity to produce the needed signal. In this case, filter LPF1 can be eliminated entirely by applying it in advance to the excitation signal. It is possible to interpolate between tables for two different striking velocities; in this case, the tables should be pre-processed to eliminate phase cancellations during cross-fade.

Assuming the first filter in Fig.9.36 is ``weakest'' at the highest hammer velocity (MIDI velocity $ 127$), that filtering can be applied to the excitation table in advance, and the first filter then becomes no computation for MIDI velocity $ 127$, and as velocity is lowered, the filter only needs to make up the difference between what was done in advance to the table and what is desired at that velocity.

Since, for most keys, only a few interaction impulses are observed, per hammer strike, in real pianos, this computational model of the piano achieves a high degree of realism for a price comparable to the cost of the strings only. The soundboard and enclosure filtering have been eliminated and replaced by a look-up table using a few read-pointers per note, and the hammer costs only one or a few low-order filters which in principle convert the interaction impulse into an accurate force pulse.

Excitation Factoring

As another refinement, it is typically more efficient to implement the highest Q resonances of the soundboard and piano enclosure using actual digital filters (see §8.8). By factoring these out, the impulse response is shortened and thus the required excitable length is reduced. This provides a classical computation vs. memory trade-off which can be optimized as needed in a given implementation. For lack of a better name, let us refer to such a resonator bank as an ``equalizer'' since it can be conveniently implemented using parametric equalizer sections, one per high-Q resonance.

A possible placement of the comb filter and equalizer is shown in Fig.9.37. The resonator/eq/reverb/comb-filter block may include filtering for partially implementing the response of the soundboard and enclosure, equalization sections for piano color variations, reverberation, comb-filter(s) for flanging, chorus, and simulated hammer-strike echoes on the string. Multiple outputs having different spectral characteristics can be extracted at various points in the processing.

Figure 9.37: Example block diagram of a more complete commuted-piano synthesis system.

Excitation Synthesis

For a further reduction in cost of implementation, particularly with respect to memory usage, it is possible to synthesize the excitation using a noise signal through a time-varying lowpass filter [440]. The synthesis replaces the recorded soundbard/enclosure sound. It turns out that the sound produced by tapping on a piano soundboard sounds very similar to noise which is lowpass-filtered such that the filter bandwidth contracts over time. This approach is especially effective when applied only to the high-frequency residual excitation after factoring out all long-ringing modes as biquads.

Coupled Piano Strings

Perhaps it should be emphasized that for good string-tone synthesis, multiple filtered delay loops should be employed for each note rather than the single-loop case so often used for simplicity (§6.12; §C.13). For piano, two or three delay loops can correspond to the two or three strings hit by the same hammer in the plane orthogonal to both the string and bridge. This number of delay loops is doubled from there if the transverse vibration planes parallel to the bridge are added; since these are slightly detuned, they are worth considering. Finally, another 2-3 (shorter) delay loops are needed to incorporate longitudinal waves for each string, and all loops should be appropriately coupled. In fact, for full accurancy, longitudinal waves and transverse waves should be nonlinearly coupled along the length of the string [391] (see §B.6.2).

Commuted Synthesis of String Reverberation

The sound of all strings ringing can be summed with the excitation to simulate the effect of many strings resonating with the played string when the sustain pedal is down [519]. The string loop filters out the unwanted frequencies in this signal and selects only the overtones which would be excited by the played string.

High Piano Key Numbers

At very high pitches, the delay-line length used in the string may become too short for the implementation used, especially when using vectorized module signals (sometimes called ``chunks'' in place of samples [355]). In this case, good results can be obtained by replacing the filtered-delay-loop string model by a simpler model consisting only of a few parallel, second-order filter sections or enveloped sinusoidal oscillators, etc. In other words, modal synthesis8.5) is a good choice for very high key numbers.

Force-Pulse Filter Design

In the commuted piano synthesis technique, it is necessary to fit a filter impulse response to the desired hammer-string force-pulse shape for each key and for each key velocity $ v_c$, as indicated in Figures 9.32 and 9.33. Such a desired curve can be found in the musical acoustics literature [488] or it can be easily generated from a good piano-hammer model (§9.3.2) striking the chosen piano string model.

For psychoacoustic reasons, it is preferable to optimize the force-pulse filter-coefficients in the frequency domain §8.6.2. Some additional suggestions are given in [467,519].

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