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Auditory environment

   

It is obvious that for speaking experiments the auditory domain plays by far the most important role of all environmental factors. At this point we assume a studio  situation, i.e. a speech recording in an anechoic chamber , and concentrate on how to provide a talker in such a situation with a reasonable impression of the room's characteristics without corrupting the speech signal at the same time. The focus is on simulation of just any room instead of a particular one; the primary goal is to avoid unwanted psychoacoustic effects due to the missing information characteristic of rooms in general.

As basic knowledge one has to understand the main acoustic components that influence the perception of one's own voice. These are:

The dilemma is that for the purpose of providing an acoustic signal to the talker without disturbing the actual speech data we need to equip him with headphones . But what type of headphone is to be chosen and how can we compensate for its effect with regard to self-perception? How to actually simulate a room, once proper headphone  compensation has been installed, and how this gives access to all kinds of acoustic subject conditioning will be discussed in Section 8.5.2.

A comprehensive elucidation of various aspects of ``Vocal Communication in Virtual Environments'' can be found in [Lehnert & Giron (1995)].

Use and compensation of headphones

 

 

Open vs. closed headphones

 

 

One could be tempted to say that closed headphones are the ideal solution to the problem of providing a talker with sound. Unfortunately, they are not.

First of all, even hearing protectors which are specially designed to be as ``closed'' as possible do not provide perfect insulation; especially at low frequencies acoustic shielding is poor. This effect is even more pronounced for closed headphones. The difference between open and closed headphones is a qualitative one and not a principled one, and the terms are somewhat misleading.

A closed headphone will change the radiation impedance at the end of the auditory channel towards the free sound field and therefore the in-head sound transmission mechanism of the voice will be severely affected. Equalisation of this effect, especially at higher frequencies, is very difficult to achieve.

Another effect caused by modified radiation impedance is that sensitivity to hearing one's own blood flow is increased. Also, sounds that are at least partly perceived by the body, e.g. one's own footsteps, are greatly modified and in general sensitivity to structure-transmitted sounds is increased.

The only benefit of using closed headphones is that external sound is shielded somewhat better than with open headphones. but in a rather quiet environment, such as an anechoic chamber , this should in any case not be of major concern. In the other direction, i.e. from the headphone to the microphone , it is necessary to be aware of the possibility of undesirable feedback (cf. Section 8.5.2).

It is always recommendable to use open headphones for the sound conditioning of a talker.

 

Using diffuse-field equalised headphones

 

 

Sound reproduction can be performed in two ways, namely by either using loudspeakers or by using headphones. Only headphones offer a means of reproducing sound in a systematic and well-defined manner. However, for that purpose a specific headphone or a specific kind of headphone must be chosen.

1. Equalising headphones
  Every headphone may be described by its frequency response.   The frequency response of a headphone can be measured by applying the headphone to an ear-like device, for instance a dummy head  with a simulated auditory channel and an ear coupler which has the same mechanical characteristics as the human ear drum, middle ear and the inner ear. Then, for all frequencies in the audible range, the ratio of the Fourier spectrum  of the sound pressure in front of the (artificial) ear drum to that of the headphone voltage can be measured. This (rather complicated) method permits prediction of the behaviour of a headphone when it emits sound to a human ear. But what should the frequency response    of a headphone look like?

The easiest method of equalising a headphone would be to equip it with a flat frequency response.   But this would result in a very unnatural sound. The reason for this is that the human outer ear, the head and the torso are a direction-dependent filter for incoming sounds. That means that the spectral shape of a sound event changes depending on its direction. Our brain is able to do inverse filtering with respect to the sound source position. The result is that when a sound source rotates around a listener's head the perceived timbre of the sound remains more or less unchanged, although the spectrum   of the sound measured at the ear drum changes dramatically. The consequence of this effect is that a headphone can only be equalised correctly for a particular condition of sound incidence.

Recently, two equalisation techniques have been developed. The first one is the so-called Free-Field Equalisation. A free-field equalised headphone  produces the same spectral distribution of sound at the ear drum of the listener as does an ideal loudspeaker placed under free-field conditions (e.g. comparable with an anechoic chamber ) in front of the listener.

The second one is the so-called Diffuse-Field Equalisation. A diffuse-field equalised headphone, when fed with white noise,   produces the same spectral distribution of sound at the ear drum of the listener as appears in a diffuse field. In a diffuse sound field the direction of incidence is evenly distributed over all directions (e.g. in a reverberation  chamber).

2. Selection of the right equalisation
  The main difference between free-field   and diffuse-field equalisation is that a free-field equalised headphone is equalised with respect to the forward direction, whereas the diffuse-field equalised headphone is equalised with respect to an average over all directions of incidence.

Most of the sound signals that are to be reproduced via headphones consist of incoming sounds from various directions. Such sound signals would require the use of a diffuse-field equalised headphone, because this type is the better choice in the sense of ``least mean error''. Even for sound sources coming from a single direction the diffuse-field equalised headphone is the better choice when the direction of incidence is not close to the forward direction.

Another point is that the free-field equalisation function varies from person to person much more than the diffuse-field equalisation function does. So an averaged diffuse-field equalisation function is valid for more people.

Both the diffuse-field equalisation function and the free-field equalisation function have been standardised, but so far, there are still differences among diffuse-field equalised and free-field equalised headphones   made by different manufacturers.

The most important reason for choosing diffuse-field equalised headphones is that recordings made for diffuse-field equalised headphones also yield good results when reproduced via loudspeakers and vice versa. Using diffuse-field equalised headphones offers the best compatibility to common recording techniques, at least in the opinion of many experts.

3. Using dummy heads and binaural simulation techniques together with headphones
   When reproducing dummy head recordings or recordings made by using binaural simulation techniques via headphones, localisation errors and coloration can only be avoided when the headphones fit well together with the recording technique. In the case of dummy head recordings, this simply means the use of free-field equalised dummy heads with free-field equalised dummy heads and the use of diffuse-field dummy heads together with diffuse-field equalised headphones. Diffuse-field equalised dummy heads (e.g. the Neumann KU 81) are mainly used for listening purposes whereas free-field equalised dummy heads (e.g. the ``Aachener Kopf'', the ``Aachen Head'', made by the Head Acoustic company) are mainly used for measurement purposes.

When making recordings using binaural simulation techniques, the catalogue of outer-ear transfer functions should also be diffuse-field equalised.

 

4. Various diffuse-field headphones
  At the moment, diffuse-field equalised headphones are available from different companies, for example:

The Stax is an electrostatic headphone and it is delivered together with a preamplifier which contains the diffuse-field equalisation. It is probably the best headphone available on the market, but it is very expensive. All headphones are open except of Sennheiser's HD 250 which is a closed type. The company ``Beyer'' sells a headphone which is delivered together with a passive network for diffuse-field equalisation.

In the SAM  Project, the AKG 240 DF was selected as a low-cost standard headphone, while the Stax SR with diffuse-field equaliser is a good choice as a highest-quality reference system .

 

Insertion-loss compensation

   

Since the subject wears a headphone, the transmission of the speech sound through the air around the head is significantly disturbed. In order to model natural self-perception of the voice, this effect has to be compensated for.

The sound field outside the head can be considered as a linear area that is free of sound sources. In such an area the sound pressure signal at any point can be reconstructed from the sound pressure signal at any other point just by knowing the correct transfer function between both points. The task of measuring this transfer function is similar to the well-known procedure of determining a Head Transfer Function: miniature microphones  are placed at the entrance of the blocked auditory channels. While speaking, the sound pressure signals at the reference point (recording microphone ) and the ear microphone  are recorded simultaneously. The magnitude of the transfer function may be obtained by averaging the energy of the short term spectra of both signals and dividing the resulting values at the ear drum by those measured at the reference point. During this procedure the phase  information is lost. A plausible phase  can be generated by calculating the minimum phase  function. The same procedure has to be done with the subject wearing the headphone such that the resulting compensation function is given to 1-l(f), where l(f) is the result of complex valued division of both transfer functions.

A sensitive matter is the choice of the reference point; if authentic compensation is desired, this point, i.e. the recording microphone , must be located as close to the mouth as possible and it must not move significantly during a recording session. The use of high-quality headsets  is therefore strongly recommended for that purpose.

Since the level of compensation is critical, the insertion-loss compensation function has to be determined for each talker individually.

A rather practical approach to the problem of insertion-loss compensation is to ask the subject to just turn the gain of his own speech signal - equalised with the Head Transfer Functions of the ear-to-mouth direction and fed back to the headphone - up to the point where it sounds as ``normal'' as possible.

 

Feedback compensation

 

Feedback compensation, i.e. the compensation for possible sound transmission from the headphones to the recording microphone , is not considered to be necessary.

If pure headphone compensation and room simulation is requested, sound is only emitted from the headphones simultaneously with the speech and thus would not degrade the signal-to-noise ratio  of the recorded signal.

Commonly used background sound, such as concurrent speakers or an underlying noise  floor that might be intended for subject conditioning should also not jeopardise the speech quality . A sensitive upper limit for those conditioning signals is considered to be a sound pressure level   of 85db.

Above this limit, audible feedback from the headphone to the recording microphone  should be taken into account during non-speaking intervals.  

Modelling of the auditory perspective

 

Spatial auralisation of sources can be performed by real-time  filtering of sound signals with the Head Transfer Function of the talker in combination with a modelling system that calculates the spatial map of secondary sources and the corresponding filter functions. In a successive stage the contributions of all secondary sources are filtered with the Head Transfer Functions for the corresponding directions of incidence. Figure 8.8 displays the auditory subsystem of the so-called SCAT-LAB that has been developed in the course of the ESPRIT  basic research project 6358 SCATIS (Spatially Coordinated Auditory/Tactile Interactive Scenario). Since SCATIS was originally designed for unidirectional simulation (passive subject) it has been augmented according to the needs for speech recordings in anechoic chambers , i.e. microphone  feedback to the DSP-network has been established, and the database has been expanded by the insertion-loss compensation   function of the headphone. 

 figure12268
Figure 8.8: Augmented auditory subsystem of the SCATIS VE generator 

Talker mirroring in the virtual room

 

With respect to the modelling of room acoustics, the subject's voice is a sound source like any other and may be modelled as such, though with a few exceptions:

  1. The direct sound component is to be handled differently. In the original room acoustics model, the sound source and the receiver are located at the same point, which is not permissible for the talker mirroring. Here, the direct sound component is produced by the speaker himself and reconstructed correctly, thanks to insertion-loss compensation described above. The consequence for room acoustic modelling is that reflections have to be rendered as normal whereas direct sound has to be omitted.
  2. The modelling of directivity with linear direction-dependent filters implies that the source or reflections is located in the far field of the source. Unfortunately, this is not the case since the point where the speech signal is picked up is very close to the mouth and therefore definitely in the near field range. Due to the linear character of the sound field, this might be corrected by a single linear filter for all directions of emission. This filter is given by the ratio of the sound pressure spectra measured at a point in the reference direction which is sufficiently far away from the sound source.
  3. With the SCAT-LAB a head-tracker is mounted to the subject which provides the system with all the necessary information to dynamically retune the position dependent parameters of the room simulation. Therefore the talker is free to move around within a recording session. Only the microphone position   has to be well defined and fixed during the recording.

Subject conditioning

 

Up to this point we have described how to properly compensate for disturbing acoustic effects due to insufficient self-perception resulting from the headphone,  or due to missing sound reflections in an anechoic chamber . In fact the technical setup for this also allows for a virtually unlimited range of acoustic conditioning of the subject. This includes ordinary noise  of defined level and spectra, or simple monaural interaction between the talker and the recording manager, as well as complex scenarios  such as dialogues in the entrance hall of a railway station with incoming trains, heavy reverberations , and concurrent speakers from different directions. Additional sounds such as these may come from a tape or they may be played in on-line. In Figure 8.8 this is summarised in the block labelled ``Audio Sources''. For later mixing and scenario  analysis it is advisable to synchronise the recorded speech with the background signal.

Recommendations on the auditory environment

 

The following recommendations are given for speech recordings in a very quiet recording environment  (e.g. an anechoic chamber ) that deprives the talking subject of most of its ``natural'' room impression. The goal is to compensate for this deficit and to decouple any acoustic feedback to the talker from the actual speech recording at the same time:

  1. Use headphones  to provide the talker with acoustic feedback/background/interaction/control in order to avoid interference with the speech recording.
  2. Choose so-called open headphones rather than closed types.  They yield less degradation of acoustic self-perception and minimise weight related discomfort.
  3. Install proper insertion-loss compensation  for the headphones.   Use high-quality headsets  for that purpose.
  4. For acoustic stimulation from directions other than the front choose diffuse-field equalised headphones. 
  5. When headphones  are used in combination with binaural simulation techniques make sure both are equalised in the same way (diffuse-field or free-field).   
  6. For proper acoustic room simulation make use of existing systems, such as the SCAT-LAB developed under ESPRIT  project No. 6358.

 



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Next: Visual environment Up: Environment characteristics Previous: Artificial vs. natural environment

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