Abstract:
A method and apparatus for having a user select from simultaneously provided audio tracks which are provided from different directions to the user. One of the tracks is provided from in front of the user and with a higher intensity than another track which is provided from a direction more to the side of the user. When the other track is selected, it is provided from in front of the user with a higher intensity, and a third track is now provided from the side of the user and at a lower intensity.
Abstract:
The present invention relates to a method for selecting auditory signal components for reproduction by means of one or more supplementary sound reproducing transducers, such as loudspeakers, placed between a pair of primary sound reproducing transducers, such as left and right loudspeakers in a stereophonic loudspeaker setup or adjacent loudspeakers in a surround sound loudspeaker setup, the method comprising the steps of (i) specifying an azimuth angle range within which one of said supplementary sound reproducing transducers is located or is to be located and a listening direction; (Ii) based on said azimuth angle range and said listening direction, determining left and right interaural level difference limits and left and right interaural time difference limits, respectively; (iii) providing a pair of input signals for said pair of primary sound reproducing transducers; (iv) pre-processing each of said input signals, thereby providing a pair of pre-processed input signals; (v) determining interaural level difference and interaural time difference as a function of frequency between said pre-processed signals; and (vi) providing those signal components of said input signals that have interauial level differences and interaural time differences in the interval between said left and right interaural level difference limits, and left and right interaural time difference limits, respectively, to the corresponding supplementary sound reproducing transducer. The invention also relates to a device for carrying out the above method and systems of such devices.
Abstract:
A method and a system for equalizing one or more loudspeaker(s), e.g. a hi-fi system, positioned in a room in order to compensate sound reproduction from the loudspeaker for an influence of the room. The method includes measuring a listening position transfer function (L) from electrical input of the loudspeaker (L1) to a sound pressure at a listening position (LP) in the room. A global transfer function (G) representing a spatial average of sound pressure level in the room generated by the loudspeaker (L1) is determined. This global transfer function (G) can either be determined as an average of two or more transfer functions measured in field points scattered across the room or it can be calculated based on an acoustic power output measured from the loudspeaker (L1) together with data regarding sound absorption properties of the room. An upper gain limit (UGL) as a function of frequency is then determined based on an inverse of the global transfer function (G). An equalizing filter (F) is then determined based on an inverse of the listening position transfer function (L), but with its gain being limited to a maximum gain in accordance with the upper gain limit (UGL). Finally, the loudspeaker (L1) is equalized with the equalizing filter (F), the filter (F) being implemented such as a minimum phase approximation by an FIR or an HR filter. Preferably, a lower gain limit (LGL) as a function of frequency is also determined as an inverse of the global transfer function (G), wherein a gain of the equalizing filter (F) is limited to a minimum gain in accordance with the lower gain limit (LGL). By use of the upper and lower gain limits (UGL, LGL) it is possible to implement a system capable of automatically designing the equalizing filter (F) with only simple tasks to perform for an operator of the system.
Abstract:
Audio processor for processing a set of input audio channels and generate a corresponding processed set of signals adapted for playback via a set of narrow-spaced loudspeakers with the purpose of providing a spatial image widening effect. The audio processor includes a cross talk canceller active only in a pre-selected frequency range, e.g. 1.5-18 kHz, and substantially in-active outside this frequency range. In addition, the audio processor includes applying substantially similar frequency weightings to the two input audio channels within the mentioned pre-selected frequency range. This frequency weighting is selected such that the processed set of signals provides a listener with a perceived timbre being substantially the same as a perceived timbre provided by the input set of audio signals. The frequency weighting is preferably based on a magnitude of an ipsi-lateral or a contra-lateral transfer function, or based on a square root of sum of squares of magnitudes of ipsi-lateral and contral-lateral transfer functions. The audio processor is advantageous since it provides a high sound quality without severe tonal coloration and with a stable spatial widening effect tolerant to listener head movements in spite of very narrow-spaced loudspeakers, such as with a listening angle of 4° or less, e.g. in a mobile phone or other handheld devices. In addition, the processor is advantageous in that it provides a high reproduction quality of both timbre and spatial aspects for normal stereo signals as well as binaural signals, including 3D spatial content in case of binaural input signals, without the need to adapt the processing to the actual input signal type.
Abstract:
A method and a system for adapting the timbre of sound output at power-up/down of a media player or stereo, where the timbre adaptation is a limitation of the bandwidth of the signal or sound, which bandwidth limitation is reduced over time to eventually provide no filtering at all.
Abstract:
The present invention relates to a method for controlling one or more loudspeakers provided in an enclosure, such as a listening room or an automobile cabin, the method comprising the steps of: (i) providing said one or more loudspeakers (2, 3, 4) with an audio input signal (5) whereby a sound field (10) is generated in the enclosure (1), and determining the corresponding acoustic power output APO(f) emitted from the one or more loudspeakers (2, 3, 4) into said enclosure (1); (ii) determining an acoustic contribution or room gain RG(f) of the enclosure (1) to the generated sound field (10); (iii) optionally determining a listening position interface LPI(f) that characterises a listener's ability to receive sound energy from a sound field at the specific place in the sound Field, in which he is located; and (iv) determining a filter characteristic as a function of the acoustic power output, the acoustic contribution or room gain RG(f) of the enclosure to the sound field in the enclosure and optionally the listening position interface between the sound field at the listening position and a listener placed at this position. The invention furthermore relates to a system for carrying out the above method and specifically to the use of the method and system for obtaining optimal audio reproduction in confined spaces such as automobile cabins.
Abstract:
Aloudspeakerbased on a dipole element (DE) with at least one diaphragm arranged to generate an acoustic dipole signal according to an electric signal, e.g. a dedicated dipole driver such as an Air Motion Transformer or a combination of two monopole drivers, e.g. dome tweeters,mounted back to back close together. An acoustic waveguide (WG) is arranged in relation to the dipole element (DE) such that a surface (S) of the acoustic waveguide (WG) is close to the at least one diaphragm of the dipole element (DE). The acoustic waveguide (WG) extends in both directions of a main axis (MA) of the dipole element (DE), thus serving to guide the acoustic dipole signals away from the dipole element (DE). Preferably, the surface (S1, S2) of the acoustic waveguide (WG) has ageneral tilt of less than 30° in relation to the main axis (MA). Thereby, a diffuse sound field is provided with only a limited requirement for housing the acoustic waveguide (WG) in the depth dimension. A smooth sound radiation for directions away from on-axis is provided, and sound radiationon-axis is highly suppressed. With these properties the loudspeaker is suited as back or surround loudspeaker in surround sound systemsto cover midrange and/or upper audio frequencies.
Abstract:
The invention relates to a way to avoid the strong reflection of the wall behind the loudspeaker is to place an acoustic reflector behind the loudspeaker driver to redirect the sound away from the wall. However, the redirected sound should also be directed away from the listening position. A direction approximately perpendicular to the wall behind the loudspeaker seems to be optimal. This can e.g. be achieved by using a triangular sound reflector as depicted on the accompanying drawings. Other shapes may be applied as well.
Abstract:
A method and a system for equalizing one or more loudspeaker(s), e.g. a hi-fi system, positioned in a room in order to compensate sound reproduction from the loudspeaker for an influence of the room. The method includes measuring a listening position transfer function (L) from electrical input of the loudspeaker (L1) to a sound pressure at a listening position (LP) in the room. A global transfer function (G) representing a spatial average of sound pressure level in the room generated by the loudspeaker (L1) is determined. This global transfer function (G) can either be determined as an average of two or more transfer functions measured in field points scattered across the room or it can be calculated based on an acoustic power output measured from the loudspeaker (L1) together with data regarding sound absorption properties of the room. An upper gain limit (UGL) as a function of frequency is then determined based on an inverse of the global transfer function (G). An equalizing filter (F) is then determined based on an inverse of the listening position transfer function (L), but with its gain being limited to a maximum gain in accordance with the upper gain limit (UGL). Finally, the loudspeaker (L1) is equalized with the equalizing filter (F), the filter (F) being implemented such as a minimum phase approximation by an FIR or an HR filter. Preferably, a lower gain limit (LGL) as a function of frequency is also determined as an inverse of the global transfer function (G), wherein a gain of the equalizing filter (F) is limited to a minimum gain in accordance with the lower gain limit (LGL). By use of the upper and lower gain limits (UGL, LGL) it is possible to implement a system capable of automatically designing the equalizing filter (F) with only simple tasks to perform for an operator of the system.
Abstract:
Audio processor for processing a set of input audio channels and generate a corresponding processed set of signals adapted for playback via a set of narrow-spaced loudspeakers with the purpose of providing a spatial image widening effect. The audio processor includes a cross talk canceller active only in a pre-selected frequency range, e.g. 1.5-18 kHz, and substantially in-active outside this frequency range. In addition, the audio processor includes applying substantially similar frequency weightings to the two input audio channels within the mentioned pre-selected frequency range. This frequency weighting is selected such that the processed set of signals provides a listener with a perceived timbre being substantially the same as a perceived timbre provided by the input set of audio signals. The frequency weighting is preferably based on a magnitude of an ipsi-lateral or a contra-lateral transfer function, or based on a square root of sum of squares of magnitudes of ipsi-lateral and contral-lateral transfer functions. The audio processor is advantageous since it provides a high sound quality without severe tonal coloration and with a stable spatial widening effect tolerant to listener head movements in spite of very narrow-spaced loudspeakers, such as with a listening angle of 4° or less, e.g. in a mobile phone or other handheld devices. In addition, the processor is advantageous in that it provides a high reproduction quality of both timbre and spatial aspects for normal stereo signals as well as binaural signals, including 3D spatial content in case of binaural input signals, without the need to adapt the processing to the actual input signal type.