Electrical audio signal processing systems and devices – Including frequency control
Reexamination Certificate
1998-05-22
2004-07-06
Mei, Xu (Department: 2644)
Electrical audio signal processing systems and devices
Including frequency control
C381S059000
Reexamination Certificate
active
06760451
ABSTRACT:
BACKGROUND OF THE INVENTION
1. Field of the Invention
This invention relates to filtering audio signals to compensate the effects of acoustic and/or electrical stages in the signal path from the original sound source to the human ear.
2. Discussion of Prior Art
In general, this signal path will include a pickup receiving the sound, and converting it to, typically, an electrical signal; signal transmission channels; signal processing (e.g. filtering, tone control or noise reduction); signal transmission, or alternatively recording on to a record carrier; signal reception or alternatively replaying from the record carrier; a further transmission link; and reconversion into an audio signal via an electro-acoustic transducer. If the transducer is a loudspeaker, the final stage in the path is transmission through an acoustic environment (typically a room) to the human ear.
Associated with each stage of the signal path is a transfer characteristic, and at various stages in the path attempts may be made to filter the signal to compensate the effects of these transfer characteristics. Compensation generally takes place at a stage in the signal path subsequent to the stages to be compensated. For example, in the case of a sound recording, the signal will be filtered at the mixing and cutting stages so as to compensate, if necessary, for the recording environment and equipment (amongst other things).
At the reproduction stage, it is nowadays common to provide a so called “graphic equalizer” comprising a plurality of band pass filters each with its own gain control, though which the signal is passed, to allow a listener to re-equalize the reproduced sound signal. The graphic equalizer is generally positioned between the record carrier reader (e.g. turntable or compact disc player) and the power amplifier driving the electro-acoustic transducer (loudspeaker).
Since such equalizers are adjusted manually, their setting is a matter for the personal taste of the listener but they can be used (and are intended for use) to compensate for large scale irregularities in the amplitude response over frequency of the electro-acoustic transducer or of the acoustic environment in which the transducer is positioned.
In fact, with modern high fidelity audio equipment, the major variations in sound reproduction quality are due to the transfer functions of the loudspeaker and of the acoustic environment in which the loudspeaker is positioned.
The loudspeaker often comprises several separate transducers responsive to different frequency ranges, the loudspeaker input signal being split into the ranges by a crossover network (generally an analogue filter), and the transducers being mounted in a cabinet. The transfer function of the loudspeaker will thus depend upon the electrical characteristics of the crossover network and of the transducers; on the relevant positions of the transducers; and on the mechanical resonances of the cabinet.
The transfer function of the acoustic environment may be visualised by considering that the signal passes though multiple paths between the loudspeaker and the human ear; as well as the direct path through the air between the two, there will generally be a path through the floor on which the loudspeaker and user stand, and reflected paths from the (at least) four walls, ceiling and floor. This leads to constructive and destructive acoustic interference and to standing wave patterns of considerable complexity within the room, so that the paths from the loudspeaker to different points in the room will have different transfer characteristics—where the room exhibits pronounced resonances, these transfer characteristics can be extremely different, with complete cancellation at some frequencies, the frequencies differing between different points. These effects are audible as colorations of the reproduced sound, and as relatively long reverberations.
It would in principle be desirable to provide a compensating filter and means for deriving the parameters of the filter such that a given sound source would be reproduced substantially identically through any loudspeaker and/or acoustic environment, so as to free the listener from the need to carefully select certain loudspeakers, and pay attention to their position within a room and to the acoustic properties of the room.
One example of a proposal to achieve exactly this is described in U.S. Pat. No. 4,458,362 and corresponding EP0094762A, in which it is proposed to provide a finite impulse response digital filter (implemented by a microcomputer and a random access memory) in the signal path preceding the loudspeaker. The coefficients of the filter are derived in an initial phase, in which a listener postions himself at his desired listening point within a room and instructs the microprocessor to generate a test signal which is propagated via the loudspeaker through the room to the listener position and picked up by a microphone carried by the listener. From the test signal and signal picked up by the microphone, the impulse response of the intervening portions of the signal path (e.g. the loudspeaker and the acoustic path through the room to that listener position) is derived and the coefficients of an FIR filter approximating the inverse transfer characteristic to that of the signal path are calculated and used in subsequent filtering.
However, this attractively simple idea suffers from major drawbacks in practice. Firstly, since the transfer characteristic of the signal path is derived to only a single listener point within a room, and since (as discussed above) the transfer characteristics of signal paths to closely spaced points in the room can have widely different transfer characteristics because of the presence of multiple room resonances, if the listener moves within the room, then the transfer characteristic derived for the filter becomes inappropriate so that, far from compensating for the effects of the room, the filter may actually further degrade the sound heard by the listener at his new position.
The disclosure of U.S. Pat. No. 4,458,362 further refers only to compensating the frequency response of elements of the signal path and ignores the phase responses of those elements. Although it is commonly thought that the human ear is relatively insensitive to phase, we have found that phase distortion, even at low levels, can be perceptually significant to a listener.
Different elements of the signal path will exhibit different phase behaviour; the behaviour of loudspeakers depends variously on the crossover network, the transducers and the cabinet dimensions. The phase response of the acoustic environment, however, can be extremely complex due to the reflection or resonances from the room boundaries. These give rise to sharp changes in the phase response of the path to a single point in the room.
Another problem is that it is possible, at some points in the room, for sound to reach a listener by a first path at a relatively low level and then by a second path at a relatively higher level; the first path could, for example, be through the floor of the room; or the first path could be a direct path from the loudspeaker through the air and the second a reflection of greater magnitude (which can occur if two reflections add up in amplitude and phase). The effect in any event is that instead of hearing a sound followed by a fainter echo, the listener hears a “pre-echo” followed by a louder sound, which is perceived as extremely unnatural.
It is relatively straightforward to cancel an echo; an IIR filter having a delay equivalent to the echo length and a loop gain equivalent to −1 times the attenuation of the echo can be used, or an FIR filter of length sufficiently long to approximate such an IIR filter can be employed with suitable tap values. However, compensating a pre-echo is considerably more difficult. A direct compensation is impossible, since the corresponding IIR filter would be unstable, and it is necessary to employ a bulk delay within the compensating filter so that the impulse response of the compensating filt
Craven Peter Graham
Gerzon Michael Anthony
Mei Xu
Nixon & Vanderhye P.C.
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