Process and device for mixing sound signals

Details Process and device for mixing sound signals Process and device for mixing sound signals

1997-12-22

2002-03-26

Isen, Forester W. (Department: 2644)

Electrical audio signal processing systems and devices

Binaural and stereophonic

Pseudo stereophonic

C381S018000, C381S061000

Reexamination Certificate

active

06363155

ABSTRACT:

CROSS-REFERENCE TO RELATED APPLICATIONS
The present application claims priority under 35 U.S.C. § 119 of Swiss Patent Application No. 2248/97 filed Sep. 24, 1997, the disclosure of which is expressly incorporated by reference herein in its entirety.
BACKGROUND OF THE INVENTION
1. Field of the Invention
The present invention relates to a process and a device for mixing sound signals.
2. Discussion of the Background Information
Devices of the type described above are generally referred to as audio mixing consoles and provide parallel processing of a plurality of sound signals. In the wake of integrating new media (HDTV, home theater, DVD), stereo technology will be replaced by multi-channel, i.e., “surround” playback processes. Surround-sound mixing consoles currently available on the market generally contain a bus matrix that is expanded to several output channels. For example, N input channels (e.g., N=8-265) are generated by mono-microphones and are processed in the individual channels, i.e., 1-N, weighted with factors, and wired to a bus bar. Control of these factors, for achieving acoustic positioning of the sound source within the room, is provided through panorama potentiometers (or “panpots”) such that an. In this context, “phantom sound sources” are created in which the listener experiences the illusion that the sound in the room is created outside the loudspeaker.
Psycho-acoustic research and experience of recent years has shown that the process mentioned above, known as “amplitude panning”, only achieves an insufficient room mapping or playback of a sound field in a room in two dimensions. Thus, the phantom sound sources can only occur on connecting lines between loudspeakers, and they are not very stable. In particular, the location of the phantom sound sources change with the specific position of the listener. However, a much more natural playback is perceived by the listener if, e.g., the following two aspects are considered:
a) Loudspeaker signals are created such that the listener receives the same relative transit time differences and frequency-dependent damping processes in the left and right ear signal, i.e., as when listening to natural sound sources. Ear signals have to be correlated in a similar fashion. At low frequencies, the transit time differences are effective for localizing sound occurrences, while at higher frequencies (e.g., >1000 Hz), amplitude (intensity) differences are for the most part effective. In conventional amplitude panning, all frequencies are substantially equally dampened and transit time differences are not considered. If one substitutes the weight factors with variable filters designed in the appropriate dimensions, both localization mechanisms can be satisfied. This process is generally referred to as a panoramic setting with the aid of filtering (i.e., “pan-filtering” ).
b) If a sound source is located in a room, the first reflections and those arriving up to a maximum of 80 msec after the direct sound aid in localizing the sound source. Distance perception particularly depends on the component of the reflections relative to the direct amount. Such reflections can be simulated in a audio mixing console or synthesized by delaying the signal several times and then assigning the signals created in this manner into different directions through the pan-filters described above.
Thus, the prior art sought to provide an audio mixing console that includes the above-mentioned features a) and b) while ensuring an affordable, i.e., a comparatively more economical, technical expenditure.
One of the first digital constructions was introduced by F. Richter and A. Persterer in “Design and Application of a Creative Audio Processor” at the 86th AES Convention in Hamburg, Germany in 1989 and published in preprint 2782. In this device, direct pairs of “head related transfer functions” (HRTF), i.e., filter functions measured with the right or left ear when a test signal is sent in a certain room direction, are used as pan-filters. An appropriate HRTF-pair is provided in accordance with an appropriate room direction to each output channel signal and to its echo that is created by delaying the signal. The stereo signals thus created are then connected to a two-channel bus bar. However, this device has the following disadvantages:
a) The playback of a single HRTF is very costly if satisfactory precision is to be achieved, i.e., non-recursive digital filters of 50°-150° and recursive digital filters of 10°-30° are required. Thus, this process occupies a significant portion of the available computer capacity of a modern digital signal processor (DSP). Further, because several echoes have to be simulated, e.g., between 5-30, for a natural playback, the entire system (with a large number of channels) becomes nearly unaffordable due to the large number of filters necessary.
b) The binaural audio mixing console only supplies a stereo signal at the output that is suitable for headphone playback While an adaptation to loudspeaker, multi-channel technology may be made by modifying the filters and increasing the number of bus bars, the expenditure would significant.
D. S. McGrath and A. Reilly introduced another device in “A Suite of DSP Tools for Creation, Manipulation and Playback of Soundfields in the Huron Digital Audi Convolution Workstation” at the 100th AES Convention held in 1996 in Copenhagen and published in the preprint 4233. In this device, the number of bus bars is reduced by using an intermediate format, independent of the number or arrangement of loudspeakers, to display the sound field. The translation to the respective output format is provided through a decoder at the bus bar output. A “B-format” decoder is suggested for reproducing the sound field, in the two-dimensional case including three channels. The signal is weighted with the factors w, x=sin &phgr; and y=cos &phgr; and transferred onto the bus bar, in which w represents the signal level and &phgr; the room direction. The B-format decoder controls the loudspeakers such that a sound field is optimally reconstructed at one point in the room in which the listener is located. However, this process has the disadvantage that the achievable localization focus is too low, i.e., neighboring and opposing loudspeakers radiate the same signal with only slight differences in the sound level. To achieve “discrete effects” an accurate high channel separation is required. In a film mix, e.g., a sound should come exactly from a certain direction. This problem can be traced back to the selected sound field format (e.g., an insufficient number of channels) or to the design of the decoder that was optimized to reproducing of the sound field, and not optimized to channel separation. A further drawback is that only a passive matrix circuit is designed in the decoder. Thus, implementation of direction-dependent “pan-filters” required at the outset would demand a significantly higher number of discretely transferred directions, as is mentioned in the following in more detail.
SUMMARY OF THE INVENTION
The present invention provides a process and device for producing the most natural sound playback over a number of loudspeakers when a different number of sound sources are present while also using a minimal amount of technical expenditure.
The present invention provides mixing 1-N sound signals to 1-M output signals by separating the sound signal from each input channel and selectively delaying the separated sound signal, selectively weighting each separated and selectively delayed sound or input signal, adding these signals to appropriate additional input signals from other input channels to one intermediate signal 1-K, and separating each separate intermediate signal into output channels 1-M, defiltering the separated intermediate signal and summing them together with the other intermediate signals. The summed-up intermediate signals together produce an output signal for a loudspeaker.
The device of the present invention for mixing sound signals from input channels E
1
-EN to output chann

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