Patent Application
20090150163
The present invention relates to methods and products for use in optimising the qualitative attributes of a multichannel sound system.
There is a disparity between the recommended location of loudspeakers for an audio reproduction system and the locations of loudspeakers that are practically possible in a given environment. Restrictions on loudspeaker placement in a domestic environment typically occur due to room shape and furniture arrangement. In an automotive environment, loudspeaker placement is usually determined by availability of space rather than optimised listening. Consequently, it may be desirable to modify signals from a pre-recorded media in order to improve on the staging and imaging characteristics of a system that has been configured incorrectly.
There is an increasing number of audio formats employing a number of different channel configurations. Until recently, only one-channel and two-channel media were available to consumers. However, the introduction of distribution media such as DVD-Video, DVD-Audio, and Super-Audio CD has made multichannel audio commonplace in domestic and automotive systems. This has meant, in many cases that there is a mismatch between the number of loudspeakers in a listening environment and the number of channels in the media. For example, it frequently occurs that a listener has only two loudspeakers but 5 channels of audio on a medium. The converse case also exists where it is desirable to play two-channel program material distributed over more than two loudspeakers. Consequently algorithms are constantly being developed in order to adapt media from one format to another. Downmix algorithms reduce the number of audio channels and upmix algorithms increase the number.
Standard recommendations for domestic and automotive sound reproduction systems state that all loudspeakers should not only be placed correctly but have matched characteristics (i.e. ITU-R BS-775). However, in typical situations, this ideal requirement is rarely met. For example, in a domestic environment, it is often the case that the built-in audio system of a television is used for the centre channel of a surround sound system. This speaker rarely matches the larger, exterior loudspeakers used for the front left and right channels. In addition, it is typical for the surround speakers to be smaller as well. Consequently, the audio signals produced by these different loudspeakers differ too much for a cohesive sound field to be created in the listening environment. Therefore, it is desirable that these differences be minimised in order to give the impression of matched loudspeaker characteristics.
The tuning of high-end automotive audio systems is increasingly concentrating on the imaging characteristics and “sound staging.” It is a challenge to achieve staging similar to that intended by the recording engineer (as is possible in a domestic situation) due to the locations of the various loudspeakers in the car. It is therefore desirable that an automatic method of choosing delay and gain parameters for the various loudspeaker drivers in an automotive environment be developed to provide a “starting point” for tuning of the car's playback system.
On the above background it is an object of the present invention to provide a method and corresponding system for reduction of the number of audio channels, whereby multiple audio channels recorded on a suitable medium (for instance 5 channels in a surround sound recording) can be played back over a lesser number of loudspeakers (for instance 2 loudspeakers in a traditional stereophonic set-up).
It is a further object of the present invention to provide a method and corresponding system for increasing the number of audio channels, whereby for instance 2 stereophonic audio channels can be played back over a larger number of loudspeakers (for instance over 5 loudspeakers as in a standard surround sound set-up).
The two procedures outlined above are referred to as a Downmix algorithm/method/system and an Upmix algorithm/method/system, respectively, as mentioned initially.
It is a specific object of the present invention to provide a method and corresponding systems by means of which the acoustic imaging characteristics and “sound staging” similar to or at least approximating that intended by the recording engineer can be achieved by the loudspeakers in a car or other confined environment.
It is a further object of the present invention to provide a method and corresponding system, which enables an end user to control the apparent “width” or “surround” content of an audio presentation.
In addition, by manipulating the locations of the virtual sound sources created by the method and system of the invention, the entire sound field can be rotated around the listener, or the virtual “sweet spot”, i.e. the optimal listening position can be moved to any desired location.
It is a still further object of the present invention to provide a method and corresponding system which can be used to simulate the differences in the frequency-dependent directivity patterns of the virtual loudspeakers (i.e. the imaginary loudspeakers simulated by the use of the method and system according to the invention) and the real loudspeakers, for instance the loudspeakers actually installed in the cabin of a vehicle.
These and other objects are according to the invention attained by a method for individually controlling the outputs from a number of pre-located loudspeakers as to magnitude and time delay of signal components emitted from these loudspeakers by conversion of a set of input signals intended for a different number and configuration of virtual loudspeakers, according to which method the pre-located and virtual loudspeakers are placed in a vector space, and where each particular pre-located loudspeaker is supplied with a signal that is obtained as the linear sum of the input signals to the virtual loudspeakers, these signals being provided with individually determined magnitude and time delays, where the magnitudes and delays are calculated by using the vectorial distances between each of the virtual loudspeakers and the particular pre-located loudspeaker.
The method and system according to the invention can be used as an algorithm for correction of loudspeaker placement, an n-to-m channel upmix algorithm or an n-to-m channel downmix algorithm.
Thus, according to the invention there is provided a method for converting a first number of signals to a second number of signals such as upmixing or downmixing n input signals to m output signals, where each of said output signals (o1, o2, o3, . . . om) is obtained as the sum of processed signals (o11, o12 . . . onm). where each of said processed signals is obtained by processing corresponding input signals (i1, i2, . . . , in) in processing means having a transfer function Hij or an impulse response hij, where the transfer function may be a function of frequency.
According to a specific embodiment of the invention, there is provided a method of the above kind for individually controlling output signals (o1, o2, o3, . . . om), which are to be provided to a number of pre-located real sound sources by conversion of a set of input signals (i1, i2, . . . in) intended for a different number and configuration of virtual sound sources, where the pre-located real sound sources and the virtual sound sources are located or represented in a vector space, and where each particular pre-located real sound source is provided with a signal (o1, o2, o3, . . . om) that has a magnitude and time delay obtained as a linear sum of at least some of said input signals intended for the virtual sound sources, and the magnitudes and delays of the signal (o1, o2, o3, . . . om) to be provided to a particular one of said real sound sources are calculated by using the vectorial distances between each of the virtual sound sources and the particular pre-located sound source.
According to the above embodiment of the invention, the signal sent to a given loudspeaker is created by summing all input channels from the playback medium with each input channel assigned an individual delay and gain. These two parameters are calculated using the relationship between the desired locations of the loudspeaker(s) and the actual location of the loudspeaker(s). For example,
d=√{square root over ((Xv−Xr)2+(Yv−Yr)+(Zv−Zr)2)}{square root over ((Xv−Xr)2+(Yv−Yr)+(Zv−Zr)2)}{square root over ((Xv−Xr)2+(Yv−Yr)+(Zv−Zr)2)}
where d is the distance between the real and virtual loudspeakers, (Xv, Yv, Zv) is the location of the virtual loudspeaker in a Cartesian coordinate system, and (Xr, Yr, Zr) is the location of the real loudspeaker. All variables are assumed to be on the same scale.
The distance between a given virtual loudspeaker and a given real loudspeaker is used to calculate a gain and delay corresponding to the gain and delay naturally incurred by propagation through that distance in a real environment. The delay can be calculated using the equation
where D is the propagation delay to be simulated, d is the calculated distance between the virtual and real loudspeakers and c is the speed of sound in air.
The gain to be applied to the signal is typically attenuation, and is also determined by the distance between the real and virtual loudspeakers. As an example, this can be calculated using the equation
where g is gain applied to the signal simulating attenuation due to distance.
Alternatively, the gain calculation could be based on sound power rather than sound pressure attenuation over distance.
The above gain/attenuation g is independent on frequency, but it is also possible according to the invention to apply a frequency-dependent g-function, i.e. g(f). By applying g(f) for instance, frequency-dependent directional characteristics of the virtual sound sources may be accounted for, and it is furthermore possible to introduce perceptual effects of the open ear transfer function of the human ear, this function being generally a function of both frequency and angle of sound incidence from the virtual sound source to the position of the listener. An illustrative example will be given in the detailed description of the invention. In this generalised case (both relating to directional characteristics of the virtual sound sources and to the incorporation of HRTF's), the function g will depend on both direction of sound incidence from a given sound source to the listening position, this direction being denoted by the vector R, and on the frequency, i.e. g as mentioned above will be replaced by (R, f).
According to the invention, there is furthermore provided an apparatus for performing a conversion or upmix/downmix operation comprising:
According to a specific embodiment of the apparatus according to the invention each of said processing means (H11, H12 . . . Hnm) comprise delay means or gain means, or both delay means and gain means, whereby each of said processed output signals (o11, o12, o13, . . . onm) will be a delayed version of the corresponding input signal or an amplified or attenuated version of the corresponding input signal or a delayed and amplified or attenuated version of the corresponding input signal.
According to a specific embodiment of the Invention, said apparatus comprises:
If the input source provides digital output signals, the series of A/D converter means mentioned under item (b) above can of course be omitted. Furthermore, if “digital” loudspeakers with digital amplifiers (for instance class-D amplifiers) are used, the D/A converter mentioned under item (h) above can also be omitted.
The present invention furthermore relates to the use of the inventive method and apparatus for supplying a set of automotive loudspeakers with signals corresponding to a home entertainment environment.
The method and apparatus according to the invention can for instance be used in domestic sound reproduction systems and automotive sound reproduction systems.
The methods can give listeners the impression that loudspeakers are correctly placed in configurations where this is not the case.
The methods can be used as a matrix that translates any desired number of channels in the distribution or playback media (i.e. 2-, 5.1-, 7.1-, 10.2-channels etc. . . . ) to any number of loudspeakers.
The methods can be used to minimise the apparent differences between loudspeakers in domestic, automotive sound systems or for sound reproduction systems in yachts.
The methods can be used to produce a suggested tuning of delay and gain parameters for instance for domestic sound systems, automotive audio systems or for sound reproduction systems in yachts.
The present invention will be more fully understood with reference to the following detailed description of embodiments of the invention and with reference to the figures.
The proposed system can be used as an n-to-m channel upmix algorithm or an n-to-m channel downmix algorithms i.e. as an algorithm for correction of loudspeaker placement.
The methods can furthermore be used as a matrix that translates any desired number of channels in the distribution or playback media (i.e. 2-, 5.1-, 7.1-, 10.2-channels etc. . . . ) to any number of loudspeakers.
The method and apparatus according to the invention can be regarded as a method/apparatus for reproducing a given number (n) of virtual sound sources (loudspeakers) by means of a different number (m) of actual physical sound sources (loudspeakers). Thus, for instance the standard loudspeaker configuration shown in
According to an embodiment of the invention the signal sent to a given loudspeaker is created by summing all input channels from a playback medium with each input channel assigned an individual delay and gain. These two parameters are calculated using the relationship between the desired locations of the virtual loudspeaker(s) and the locations of the actual loudspeaker(s). For example,
d=√{square root over ((Xv−Xr)2+(Yv−Yr)2+(Zv−Zr)2)}{square root over ((Xv−Xr)2+(Yv−Yr)2+(Zv−Zr)2)}{square root over ((Xv−Xr)2+(Yv−Yr)2+(Zv−Zr)2)}
where d is the distance between the real and virtual loudspeakers, (Xv, Yv, Zv) is the location of the virtual loudspeaker in a Cartesian coordinate system, and (Xr, Yr, Zr) is the location of the real loudspeaker. All variables are assumed to be on the same scale.
The distance between a given virtual loudspeaker and a given real loudspeaker is used to calculate a gain and delay corresponding to the gain and delay naturally incurred by propagation through that distance in a real environment. The delay can be calculated using the equation
where D is the propagation delay to be simulated, d is the calculated distance between the virtual and real loudspeakers and c is the speed of sound in air.
The gain to be applied to the signal is typically attenuation, and is also determined by the distance between the real and virtual loudspeakers. As an example, this can be calculated using the equation
where g is the gain applied to the signal simulating attenuation due to distance.
An apparatus corresponding to the situation shown in
With reference to
With reference to
Referring to
Referring to
With reference to
In order to program the apparatus, X, Y, Z coordinates 63, 64 of the real loudspeakers 55, 56 and X, Y, Z coordinates I, II, III, IV, V of the virtual loudspeakers 1′, 2′, 3′, 4′, 5′ are entered by means of a suitable user interface, for instance by the touch screen device 61 schematically shown in
With reference to
ΔHRTF4=ΔHRTF5=HRTF(β)−HRTF(α)
where it is assumed that the head-related transfer functions from the virtual loudspeakers 4′ and 5′ to the listener 71 are identical, which in principle will be true in this case, as the set-up is symmetrical with respect to the median plane through the listener 71 indicated by 72 in
As mentioned above in connection with
| Number | Date | Country | Kind |
|---|---|---|---|
| PA 2004 -01816 | Nov 2004 | DK | national |
| Filing Document | Filing Date | Country | Kind | 371c Date |
|---|---|---|---|---|
| PCT/IB05/53830 | 11/21/2005 | WO | 00 | 5/21/2007 |
