公开(公告)号：WO2006076926A3

公开(公告)日：2006-10-12

申请号：PCT/DK2006000234

申请日：2006-04-28

Applicant: AM3D AS ,  PEDERSEN JAN ABILDGAARD

Inventor： PEDERSEN JAN ABILDGAARD

IPC: H04S1/00

CPC classification number: H04S1/002 ,  H04S2400/01 ,  H04S2420/01

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 translation: 音频处理器，用于处理一组输入音频声道并产生适合于通过一组窄间距扬声器重放的相应处理信号集合，目的是提供空间图像加宽效果。 音频处理器包括仅在预选频率范围内激活的串音消除器，例如， 1.5-18kHz，并且在该频率范围之外基本处于非活动状态。 另外，音频处理器包括在所提及的预选频率范围内对两个输入音频声道施加基本相似的频率加权。 该频率加权被选择为使得经处理的一组信号向收听者提供与由输入音频信号集提供的感知音质基本相同的感知音质。 频率加权优选地基于同侧或对侧传递函数的幅度或基于同侧和对侧 - 横向传递函数的平方和的平方根。 音频处理器是有利的，因为它提供了高音质而没有严重的音调着色，并且具有对听众头部运动的宽容的稳定空间，尽管是非常窄间距的扬声器，例如具有4°或更小的收听角度， 在手机或其他手持设备中。 此外，处理器的优点在于，它为正常的立体声信号以及双耳信号（包括在双耳输入信号的情况下的3D空间内容）提供了音质和空间方面的高重放质量，而无需使处理适应于 实际的输入信号类型。

公开(公告)号：WO2007076863A8

公开(公告)日：2008-12-24

申请号：PCT/DK2006000724

申请日：2006-12-19

Applicant: SLH AUDIO AS ,  PEDERSEN JAN ABILDGAARD

Inventor： PEDERSEN JAN ABILDGAARD

IPC: H04S7/00 ,  H04S3/00

CPC classification number: H04S3/00 ,  H04S7/00

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 translation: 一种用于均衡一个或多个扬声器的方法和系统，例如， 高保真系统，定位在房间中，以便补偿来自扬声器的声音再现对于房间的影响。 该方法包括从扬声器（L1）的电输入到室内听音位置（LP）的声压测量收听位置传递函数（L）。 确定表示由扬声器（L1）生成的房间中的声压级的空间平均的全局传递函数（G）。 该全局传递函数（G）可以被确定为分散在房间中的场点中测量的两个或更多个传递函数的平均值，或者可以基于从扬声器（L1）测量的声功率输出以及关于 房间的吸音性能。 然后，基于全局传递函数（G）的倒数来确定作为频率的函数的上限增益限制（UGL）。 然后，基于收听位置传递函数（L）的倒数确定均衡滤波器（F），但是根据上限增益限制（UGL）将其增益限制为最大增益。 最后，扬声器（L1）与均衡滤波器（F）相等，滤波器（F）被实现为通过FIR或HR滤波器的最小相位逼近。 优选地，作为频率的函数的较低增益极限（LGL）也被确定为全局传递函数（G）的倒数，其中均衡滤波器（F）的增益被限制为根据较低的 增益限制（LGL）。 通过使用上限和下限增益限制（UGL，LGL），可以实现一个能够自动设计均衡滤波器（F）的系统，只需要为系统的操作者执行简单的任务即可。