Apparatus and method for providing individual sound regions

Abstract

To provide an apparatus and a method for providing individual sound zones that are solved by computer programs.SOLUTION: There is provided an apparatus for generating a plurality of loudspeaker signals from two or more audio source signals. Each of the two or more audio source signals is reproduced in one or more of two or more sound zones, and at least one of the two or more audio source signals is not reproduced in at least one of the two or more sound zones. The apparatus comprises: an audio preprocessor 110 for modifying each of two or more initial audio signals to obtain two or more preprocessed audio signals; and a filter 140 for generating the plurality of loudspeaker signals depending on the two or more preprocessed audio signals.SELECTED DRAWING: Figure 1

Description

Technical Field

[0001] The present invention relates to audio signal processing, and more particularly, to an apparatus and method for providing individual sound regions.

Background Art

[0002] Playing different acoustic scenes in a plurality of acoustic regions located close to each other without an acoustic barrier is a well-known task in audio signal processing, and this is often referred to as multi-zone playback (see [1]). From a technical perspective, multi-zone playback is closely related to speaker beamforming or spot forming (see [2]) when considering near-field scenarios where the openings of the speaker array may surround the listener.

[0003] A problem in multi-zone playback scenarios can be, for example, providing substantially different acoustic scenes (e.g., different music or audio content of different movies) to listeners occupying individual sound regions.

[0004] TIFF0007844407000001.tif31151

[0005] When playing multiple signals in a real-world enclosure, complete separation is impossible because sound waves cannot be stopped without an acoustic barrier. Therefore, there is always crosstalk between individual sound regions occupied by individual listeners.

[0006] TIFF0007844407000002.tif59157TIFF0007844407000003.tif58157

[0007] TIFF0007844407000004.tif32154

[0008] TIFF0007844407000005.tif52149

[0009] An approach to overcome this is to use directional loudspeakers, whose directivity is typically higher at higher frequencies (see

[35] :JP 5345549 and

[21] :US 2005 / 0190935 A1). Unfortunately, this method is not always effective. It is only suitable for very high frequencies (see [1]).

[0010] Another approach is to utilize a loudspeaker array in combination with appropriate pre-filters for personalized audio playback.

[0011] Figure 4 shows a minimal example of multi-zone reproduction using an array. Specifically, Figure 4 shows a basic configuration with two signal sources 211, 212, two speakers, and two regions 221, 222. The example in Figure 4 is a placeholder for more complex scenarios that may occur in actual applications.

[0012] TIFF0007844407000006.tif25151

[0013] Figure 6 shows a typical signal model for multi-zone playback using an array. The signal source 610, pre-filter 615, impulse response 417, and sound regions 221,222 are shown.

[0014] TIFF0007844407000007.tif138158

[0015] Here, the expression for equation (3) is: TIFF0007844407000008.tif37156

[0016] TIFF0007844407000009.tif82157

[0017] Each audio signal has a sound region where the signal should be reproduced, the so-called "bright region." At the same time, there is a region where the individual signal should not be reproduced, the "dark region."

[0018] For example, in Figure 3, signal source 211 is reproduced in sound region 221 but not in sound region 222. Furthermore, in Figure 3, signal source 212 is reproduced in sound region 222 but not in sound region 221.

[0019] TIFF0007844407000010.tif47152TIFF0007844407000011.tif119157

[0020] Figure 5 shows examples of playback levels in bright and dark regions, along with the resulting acoustic contrast. Specifically, Figure 5 shows examples of playback levels in bright and dark regions in (a) and the resulting acoustic contrast in (b).

[0021] TIFF0007844407000012.tif31156

[0022] TIFF0007844407000013.tif17150

[0023] Difficulties arise when directional audio playback is performed.

[0024] Some of the approaches described above attempt to achieve multi-zone reproduction through directional acoustic radiation. Such approaches face the following major physical challenges.

[0025] TIFF0007844407000014.tif82157

[0026] Since acoustic waves follow the same wave equation, this rule is also applicable to acoustic waves. Ultimately, due to technical reasons, the size of the speaker diaphragm or the aperture of the horn is limited, which means the lower limit of the frequency at which directional reproduction can be effectively achieved. Furthermore, regardless of the size of individual loudspeakers, the same applies to a loudspeaker array, which is the overall dimension of the loudspeaker array. Different from the drivers of individual loudspeakers, the size of the array is mainly restricted for technical reasons although it is also related to economy.

[0027] TIFF0007844407000015.tif64152 The solution has an effective frequency limit.

[0028] Furthermore, an enclosure that needs to create multiple sound regions may affect the radiation pattern itself that is achieved. In the case of higher frequencies, a large enclosure, and straight walls, models can be found that analytically consider the geometry of the enclosure in the design of a directional loudspeaker or prefilter for speaker array reproduction. However, this is no longer possible when the enclosure shows (general) curvature, when obstacles of arbitrary shape are placed inside the enclosure, or when the dimensions of the enclosure are on the order of the wavelength. Such settings exist, for example, inside a vehicle and will be referred to as complex settings below. Under such circumstances, it is very difficult to excite a sound field controlled by a directional speaker or an electrically steered array because the sound reflected from the enclosure cannot be accurately modeled. Under such conditions, even an omnidirectional individually driven loudspeaker can effectively exhibit an uncontrolled directional pattern. An omnidirectional individually driven loudspeaker can effectively exhibit an uncontrolled directional pattern.

[0029] Some of the prior art documents relate to (cross) signal-dependent gain control.

[0030] U.S. Patent Application Publication No. 2005 / 0152562 (see [8]) relates to in-car surround sound reproduction using different operating modes associated with different loudness patterns and different equalization patterns on individual seats.

[0031] U.S. Patent Application Publication No. 2013 / 170668 (see [9]) describes mixing an announcement sound into an entertainment signal. The mix of both signals is separate for each of the two regions.

[0032] U.S. Patent Application Publication No. 2008 / 0071400 (see

[10] ) discloses source or content information-dependent signal processing that considers two different signals to mitigate the driver becoming “acoustically overloaded.”

[0033] U.S. Patent Application Publication No. 2006 / 0034470 (see

[11] ) relates to equalization, compression, and “mirror image” equalization for reproducing sound in high-noise conditions with improved quality.

[0034] U.S. Patent Application Publication No. 2011 / 0222695 (see

[12] ) discloses speech compression of a subsequently played audio track, taking into account ambient noise and a psychoacoustic model.

[0035] U.S. Patent Application Publication No. 2009 / 0232320 (see

[13] ) describes a compression that is louder than an entertainment program and involves user interaction.

[0036] U.S. Patent Application Publication No. 2015 / 0256933 (see

[14] ) discloses balance levels for telephone and entertainment content to minimize audio leakage of content.

[0037] U.S. Patent No. 6,674,865 (see

[15] ) relates to automatic gain control for hands-free telephones.

[0038] German Patent Application Publication No. 3045722 (see

[16] ) discloses parallel compression for noise levels and level increases for announcements.

[0039] Other prior art documents relate to multizone reproduction.

[0040] U.S. Patent Application Publication No. 2012 / 0140945 (see

[17] ) relates to an implementation of an explicit sound domain. High frequencies are reproduced by a speaker, and low frequencies use constructive and destructive interference by manipulating amplitude, phase, and delay. To determine how the amplitude, phase, and delay must be manipulated,

[17] proposes a special technique, the "Tan Theta" method, or solving an eigenvalue problem.

[0041] U.S. Patent Application Publication No. 2008 / 0273713 (see

[18] ) discloses a sound area including speaker arrays positioned near each seat, with loudspeaker arrays explicitly assigned to each area.

[0042] U.S. Patent Application Publication No. 2004 / 0105550 (see

[19] ) relates to a non-directional sound region near the head, away from the listener.

[0043] U.S. Patent Application Publication No. 2006 / 0262935 (see

[20] ) explicitly relates to the personal sound domain.

[0044] U.S. Patent Application Publication No. 2005 / 019035 (see

[21] ) relates to a headrest or seatback cloud speaker for personalized playback.

[0045] U.S. Patent Application Publication No. 2008 / 0130922 (see

[22] ) discloses an implementation of a directional speaker near the front seats, an omnidirectional speaker near the rear seats, and a healthy region using signal processing to prevent leakage between the front and rear.

[0046] U.S. Patent Application Publication No. 2010 / 0329488 (see

[23] ) describes a sound area of ​​a vehicle comprising at least one speaker and one microphone associated with each area.

[0047] German Patent Application Publication No. 102014210105 (see

[24] ) relates to sound domains achieved by binaural playback using crosstalk cancellation (between ears) and reduction of crosstalk between regions.

[0048] U.S. Patent Application Publication No. 2011 / 0286614 (see

[25] ) discloses a healthy area involving crosstalk cancellation and binaural playback based on head tracking.

[0049] U.S. Patent Application Publication No. 2007 / 0053532 (see

[26] ) discloses a headrest loudspeaker.

[0050] U.S. Patent Application Publication No. 2013 / 0230175 (see

[27] ) relates to a sound domain that explicitly uses a microphone.

[0051] International Publication No. 2016 / 008621 (see

[28] ) discloses a head and torso simulator.

[0052] Further prior art literature relates to directional reproduction.

[0053] U.S. Patent Application Publication No. 2008 / 0273712 (see

[29] ) discloses a directional loudspeaker mounted on a vehicle seat.

[0054] U.S. Patent No. 5,870,484 (see

[30] ) describes stereo playback using directional loudspeakers.

[0055] U.S. Patent No. 5,809,153 (see

[31] ) relates to using three loudspeakers as an array, with three directions pointed as the circuit.

[0056] U.S. Patent Application Publication No. 2006 / 0034467 (see

[32] ) discloses a sound region relating to the excitation of a headliner by a special transducer.

[0057] U.S. Patent Application Publication No. 2003 / 0103636 (see

[33] ) relates to a headrest array that generates a sound field in the listener's ear, including personalized playback and silencing, and silencing. do.

[0058] U.S. Patent Application Publication No. 2003 / 0142842 (see

[34] ) relates to a headrest speaker.

[0059] Japanese Patent No. 5345549 (see

[35] ) refers to a parametric speaker for the front seat.

[0060] U.S. Patent Application Publication No. 2014 / 0056431 (see

[36] ) relates to directional reproduction.

[0061] U.S. Patent Application Publication No. 2014 / 0064526 (see

[37] ) relates to generating binaural and localized audio signals for a user.

[0062] U.S. Patent Application Publication No. 2005 / 0069148 (see

[38] ) discloses the use of a loudspeaker in a headlining in response to delay.

[0063] U.S. Patent No. 5,081,682 (see

[39] ), German Utility Model Registration No. 9015454 (see

[40] ), U.S. Patent No. 5,550,922 (see

[41] ), U.S. Patent No. 5,434,922 (see

[42] ), U.S. Patent No. 6,078,670 (see

[43] ), U.S. Patent No. 6,674,865 (see

[44] ), German Patent Application Publication No. 10052104 (see

[45] ), and U.S. Patent Application Publication No. 2005 / 0135635 (see

[46] ) relate to gain adaptation or spectral modification of a signal from measured or estimated ambient noise, such as velocity.

[0064] German Patent Application Publication No. 10242558 (see

[47] ) discloses antiparallel volume control.

[0065] U.S. Patent Application Publication No. 2010 / 0046765 (see

[48] ) and German Patent Application Publication No. 102010040689 (see

[49] ) relate to optimized crossfading between later played audio scenes.

[0066] U.S. Patent Application Publication No. 2008 / 0103615 (see

[50] ) describes event-dependent panning variations.

[0067] U.S. Patent No. 8,190,438B1 (see

[51] ) describes spatial rendering adjustments that depend on signals in an audio stream.

[0068] International Publication No. 2007 / 098916 (see

[52] ) describes the playback of a warning sound.

[0069] U.S. Patent Application Publication No. 2007 / 0274546 (see

[53] ) determines which musical pieces can be performed in combination with other musical pieces.

[0070] U.S. Patent Application Publication No. 2007 / 0286426 (see

[54] ) describes mixing one audio signal (e.g., telephone) with another audio signal (e.g., music).

[0071] Some prior art documents describe audio compression and gain control.

[0072] U.S. Patent No. 5,018,205 (see

[55] ) relates to band-selective adjustment of gain in the presence of ambient noise.

[0073] U.S. Patent No. 4,944,018 (see

[56] ) discloses a speed-controlled amplifier.

[0074] German Patent Application Publication No. 10351145 (see

[57] ) relates to frequency-dependent amplification for overcoming frequency-dependent thresholds.

[0075] Several prior art documents relate to noise cancellation.

[0076] Japanese Patent Publication No. 2003-255954 (see

[58] ) discloses an active noise reduction method using a speaker placed near the listener.

[0077] U.S. Patent No. 4,977,600 (see

[59] ) discloses the attenuation of pick-up noise for individual seats.

[0078] U.S. Patent No. 5,416,846 (see

[60] ) describes active noise cancellation using adaptive filters.

[0079] Further prior art literature relates to array beamforming for speech.

[0080] U.S. Patent Application Publication No. 2007 / 0030976 (see

[61] ) and Japanese Patent Publication No. 2004-363696 (see

[62] ) disclose array beamforming for sound reproduction, delay, and total beamforming.

[0081] It would be highly desirable if an improved concept were provided that offered multi-zone reproduction within a sufficient range of the audible frequency spectrum. [Overview of the project] [Problems that the invention aims to solve]

[0082] The object of the present invention is to provide an improved concept for audio signal processing. The object of the present invention is solved by the apparatus described in claim 1, the method described in claim 16, and the computer program described in claim 17. [Means for solving the problem]

[0083] An apparatus is provided for generating multiple speaker signals from two or more sound source signals. Each of the two or more sound source signals is reproduced in one or more of two or more sound regions, and at least one of the two or more sound source signals is not reproduced in at least one of the two or more sound regions. The apparatus includes an audio preprocessor configured to modify each of the two or more initial audio signals in order to obtain two or more preprocessed audio signals. Furthermore, the apparatus includes a filter configured to generate multiple speaker signals in accordance with the two or more preprocessed audio signals. The audio preprocessor is configured to use two or more sound source signals as two or more initial audio signals, or to generate initial audio signals for each of the two or more initial audio signals by modifying the sound source signals. Furthermore, the audio preprocessor is configured to modify each of the two or more initial audio signals depending on the signal power of the two or more initial audio signals or the loudness of another initial audio signal. The filter is configured to generate multiple speaker signals depending on which of the two or more sound regions the two or more sound source signals should be reproduced in, and depending on which of the two or more sound regions the two or more sound source signals should be reproduced in.

[0084] Furthermore, a method is provided for generating multiple speaker signals from two or more sound source signals. Each of the two or more sound source signals is reproduced in one or more of two or more sound regions, and at least one of the two or more sound source signals is not reproduced in at least one of the two or more sound regions. This method is - Modify each of two or more initial audio signals to obtain two or more pre-processed audio signals. And: - Generates multiple speaker signals based on two or more pre-processed audio signals.

[0085] Two or more sound source signals are used as two or more initial audio signals, or, for each of the two or more sound source signals, the initial audio signals of the two or more initial audio signals are generated by modifying the sound source signals. Each initial audio signal of the two or more initial audio signals is modified according to the signal power or loudness of another initial audio signal among the two or more initial audio signals. Multiple speaker signals are generated depending on which of the two or more sound regions the two or more sound source signals are in, and in the two or more sound regions, two or more sound source signals are not reproduced.

[0086] Furthermore, computer programs are provided, each of which is configured to implement one of the methods described above when executed on a computer or signal processor.

[0087] Some embodiments provide signal-dependent level changes that reduce perceived sound leakage when using a scale for directional reproduction of independent entertainment signals.

[0088] In this embodiment, an optional combination of differential reproduction concepts for different frequency bands is employed.

[0089] Optionally, some embodiments use a least-squares optimized FIR filter (FIR = finite impulse resonance) based on a once-measured impulse response. Details of some embodiments are described below when the pre-filter according to the embodiment is described.

[0090] Some embodiments may be used in automotive scenarios, but are not limited to such scenarios.

[0091] Some embodiments relate to the concept of providing individual audio content to listeners occupying the same enclosure without the use of headphones or the like. In particular, these embodiments differ from the latest technologies by employing a smart combination of different playback approaches with signal-dependent preprocessing that achieves a large perceived acoustic contrast while maintaining a high level of audio quality.

[0092] Some embodiments provide filter designs.

[0093] Some embodiments utilize additional signal-dependent processing.

[0094] Embodiments of the present invention will be described in more detail below with reference to the drawings. [Brief explanation of the drawing]

[0095] [Figure 1] This describes an apparatus for generating multiple speaker signals from two or more sound source signals according to one embodiment. [Figure 2] It demonstrates ideal multi-zone playback. [Figure 3] In reality, it shows the playback of multiple signals. [Figure 4] This shows a minimal example of multi-zone regeneration using an array. [Figure 5]An example of the reproduction levels of the bright and dark regions is shown in (a), and the resulting acoustic contrast is shown in (b). [Figure 6] This shows a typical signal model for multi-zone playback using an array. [Figure 7] This demonstrates multi-zone regeneration using an array according to one embodiment. [Figure 8] An example of an implementation of an audio preprocessing device according to one embodiment is shown. [Figure 9] (a) shows the acoustic contrast achieved by different reproduction methods, and (b) shows an exemplary design of a demultiplexer according to an embodiment that shows the selected amplitude response of the audio crossover. [Figure 10] This illustrates an exemplary design of a demultiplexer according to an embodiment, where (a) shows the acoustic contrast achieved by a particular reproduction method, and (b) shows the selected amplitude response of a spectral shaping filter. [Figure 11] An exemplary loudspeaker setup within an enclosure according to one embodiment is shown. [Modes for carrying out the invention]

[0096] Figure 1 shows an apparatus for generating multiple speaker signals from two or more sound source signals according to one embodiment. Each of the two or more sound source signals is reproduced in one or more of two or more sound regions, and at least one of the two or more sound source signals is not reproduced in at least one of the two or more sound regions.

[0097] The device includes an audio preprocessor 110 configured to modify each of two or more initial audio signals in order to obtain two or more preprocessed audio signals. Furthermore, the device includes a filter 140 configured to generate multiple speaker signals in response to the two or more preprocessed audio signals. The audio preprocessor 110 is configured to use two or more sound source signals as two or more initial audio signals, or the audio preprocessor 110 is configured to generate the initial audio signals of the two or more initial audio signals for each of the two or more sound source signals by modifying the sound source signal. Furthermore, the audio preprocessor 110 is configured to modify each of the two or more initial audio signals according to the signal power of the two or more initial audio signals or the loudness of the other initial audio signals.

[0098] The filter 140 is configured to generate multiple speaker signals depending on which of the two or more sound regions in which two or more sound source signals should be reproduced, and depending on which of the two or more sound regions in which two or more sound sources should not be reproduced.

[0099] While current technological approaches can achieve considerable acoustic contrast, the contrast achieved by prior art methods is typically insufficient to provide multiple unrelated acoustic scenes to the inhabitant within the same enclosure, and high-quality audio reproduction is always required.

[0100] The acoustic contrast perceived by the listener is improved, which depends on, but is not identical to, the acoustic contrast as defined by equation (14) above. Rather than maximizing the acoustic energy contrast, the acoustic contrast perceived by the listener is improved. An increase in the contrast must be achieved. Perceived acoustic contrast is called subjective acoustic contrast, and acoustic energy contrast is referred to below as objective acoustic contrast. Some embodiments use means to facilitate directional sound reproduction and means to shape acoustic leakage to make sound leakage less noticeable.

[0101] In addition to Figure 1, the apparatus in Figure 7 further comprises two (optional) band dividers 121, 122 and four (selective) spectral shapers 131, 132, 133, 134.

[0102] According to some embodiments, the apparatus may further include two or more band dividers 121, 122 configured to band divide two or more preprocessed audio signals into a plurality of band-divided audio signals. The filter 140 may be configured to generate a plurality of speaker signals according to the plurality of band-divided audio signals.

[0103] In some embodiments, the apparatus further comprises, for example, one or more spectral shapers 131, 132, 133, 134, and is configured to modify the spectral envelope of one or more of a plurality of band-divided audio signals in order to obtain one or more spectrally shaped audio signals.

[0104] TIFF0007844407000016.tif25152

[0105] Figure 7 shows two signal sources, and two independent signals are supplied to the “preprocessing” stage. In some embodiments, this preprocessing stage can perform parallel processing (i.e., no mixing) for both signals. Unlike other processing steps, this processing step does not constitute an LT1 system (linear time-invariant system). Instead, this processing block determines the time-varying gain of all processed sound source signals so that the difference in playback levels is small. The rationale behind this is that the acoustic leakage in each region always depends linearly on the scenes reproduced in the other regions. At the same time, intentionally reproduced scenes can shield acoustic leakage. Therefore, perceived acoustic leakage is proportional to the level difference between the intentionally reproduced scenes in each region. As a result, reducing the level difference of the reproduced scenes reduces perceived acoustic leakage and, therefore, increases subjective acoustic contrast. The preprocessing is described below.

[0106] TIFF0007844407000017.tif89151

[0107] As mentioned above, the means for directional reproduction applied later always exhibit a certain degree of leakage from one region to another. This leakage can be measured as a breakdown of acoustic contrast between regions. In complex settings, these breakdowns can occur at multiple points in the frequency spectrum for each of the assumed directional reproduction methods, posing a significant obstacle to the application of these methods. It is well known that some degree of timbral variation is tolerable. These degrees of freedom can be used to attenuate contrast-critical frequency bands.

[0108] Therefore, the (optional) spectral shapers 131, 132, 133, and 134 are designed so that the signal to be reproduced later is attenuated in these parts of the frequency spectrum, resulting in a lower acoustic contrast. Unlike demultiplexers, spectral shapers are intended to alter the timbre of the reproduced sound. Furthermore, this processing step may also include delay and gain so that the deliberately reproduced acoustic scenes can spatially mask acoustic leakage.

[0109] TIFF0007844407000018.tif51151

[0110] Other embodiments employ the above approach by operating with a calculated impulse response. In certain embodiments, the impulse response is calculated to represent the free-field impulse response from the speaker to the microphone.

[0111] In further embodiments, the above approach is employed by operating with a calculated impulse response obtained using an image source model of the enclosure.

[0112] Note that the impulse response is measured only once, so that a microphone is not needed during operation. Unlike ACC, the pressure matching approach defines a predetermined magnitude and phase in each bright region. This results in high reproduction quality. Conventional beamforming techniques are also suitable when high frequencies need to be reproduced.

[0113] TIFF0007844407000019.tif6148TIFF0007844407000020.tif50152

[0114] The embodiments of the present invention will be described in more detail below.

[0115] First, the preprocessing according to the embodiment will be described. In particular, an implementation of the block shown by "Preprocessing" in Figure 7 will be presented. For the sake of understanding, the following explanation will focus on only one monaural signal per region. However, generalization to multi-channel signals is straightforward. Therefore, some embodiments will show multi-channel signals for each region.

[0116] TIFF0007844407000021.tif26152

[0117] TIFF0007844407000022.tif7152

[0118] TIFF0007844407000023.tif13152

[0119] TIFF0007844407000024.tif19152

[0120] TIFF0007844407000025.tif39152TIFF0007844407000026.tif39152

[0121] Signal normalization has already reduced their relative level differences. However, this is typically insufficient for the intended effect, because power estimates are long-term, while level fluctuations in a typical acoustic scene are rather short-term processes. Below, we will explain how the reduction of the relative power differences of individual signals in the short term constitutes the main objective of the preprocessing block.

[0122] TIFF0007844407000027.tif14152

[0123] TIFF0007844407000028.tif19152

[0124] TIFF0007844407000029.tif20152

[0125] TIFF0007844407000030.tif20152

[0126] These signals are, for example, TIFF0007844407000031.tif31159TIFF0007844407000032.tif21152

[0127] TIFF0007844407000033.tif7152

[0128] TIFF0007844407000034.tif26152

[0129] TIFF0007844407000035.tif58151

[0130] According to some embodiments, the audio preprocessor 110 can be configured to modify each of the two or more initial audio signals in accordance with the signal power or loudness of another initial audio signal among the two or more initial audio signals, for example, by including determining a gain for the initial audio signal and applying the gain to the initial audio signal. Furthermore, the audio preprocessor 110 may be configured to determine the gain in accordance with, for example, a ratio between a first value and a second value, where the ratio is the ratio between the signal power of the other initial audio signal of the two or more initial audio signals and the signal power of the initial audio signal, or where the ratio is the ratio between the loudness of the other initial audio signal of the two or more initial audio signals and the loudness of the initial audio signal as the second value.

[0131] In some embodiments, the audio preprocessor 110 may be configured to determine the gain according to a function that increases monotonically by the ratio between a first value and a second value, for example.

[0132] TIFF0007844407000036.tif14150

[0133] TIFF0007844407000037.tif20151

[0134] TIFF0007844407000038.tif32158

[0135] TIFF0007844407000039.tif13154

[0136] TIFF0007844407000040.tif51152

[0137] TIFF0007844407000041.tif7152

[0138] TIFF0007844407000042.tif57159

[0139] TIFF0007844407000043.tif7159TIFF0007844407000044.tif20155

[0140] TIFF0007844407000045.tif12152

[0141] TIFF0007844407000046.tif33152

[0142] TIFF0007844407000047.tif25158

[0143] TIFF0007844407000048.tif39154

[0144] TIFF0007844407000049.tif54157TIFF0007844407000050.tif63154

[0145] TIFF0007844407000051.tif13154

[0146] TIFF0007844407000052.tif96157

[0147] TIFF0007844407000053.tif6157

[0148] TIFF0007844407000054.tif20154

[0149] TIFF0007844407000055.tif53153

[0150] TIFF0007844407000056.tif48149

[0151] TIFF0007844407000057.tif47154

[0152] TIFF0007844407000058.tif60156

[0153] TIFF0007844407000059.tif63154

[0154] TIFF0007844407000060.tif102154

[0155] TIFF0007844407000061.tif39154

[0156] The following describes further features of the pretreatment according to the embodiment.

[0157] TIFF0007844407000062.tif19152

[0158] TIFF0007844407000063.tif25152

[0159] According to one embodiment, the power estimator can be replaced with a loudness estimator, such as the one described in ITU-R Recommendation BS.1770-4. This improves playback quality because the perceived loudness is well matched by this model.

[0160] TIFF0007844407000064.tif19152

[0161] TIFF0007844407000065.tif19152

[0162] TIFF0007844407000066.tif32152

[0163] The desired frequency response of the input-output path is, for example, a flat frequency response in the passband. This allows for a bandpass with high attenuation in the stopband. The boundary between the passband and stopband is selected according to the frequency range in which the playback means connected to each output can achieve sufficient acoustic contrast between their respective acoustic bands.

[0164] Figure 9 shows exemplary designs of one or more demultiplexers according to an embodiment, where (a) shows the acoustic contrast achieved by different reproduction methods, and (b) shows the selected amplitude response of the audio crossover. In particular, Figure 9 shows an exemplary design of the filter amplitude response with respect to the achieved acoustic contrast.

[0165] As can be seen in Figure 9, the spectral shaper may be configured, for example, to modify the spectral envelope of the audio signal in response to acoustic contrast.

[0166] Various concepts can be employed to implement one or more band dividers in practice. For example, some embodiments use FIR filters, others use IIR filters, and further embodiments use analog filters. Possible concepts for implementing a demultiplexer can be any concept shown in general literature on the topic, for example.

[0167] Some embodiments may include, for example, a spectral shaping device for performing spectral shaping. When spectral shaping is performed on an audio signal, the spectral envelope of the audio signal may be modified, for example, to obtain a spectrally shaped audio signal.

[0168] TIFF0007844407000067.tif32151

[0169] However, the final frequency response of a spectral filter is completely different from that of an equalizer. The spectral filter is designed to take into account the maximum spectral distortion acceptable to the listener, and to attenuate frequencies known to produce acoustic leakage.

[0170] The rationale behind this is that human perception is differently sensitive to spectral distortions in an acoustic scene at specific frequencies, depending on the excitation of surrounding frequencies and whether the distortion is attenuated or amplified.

[0171] For example, if a narrow-bandwidth notch filter is applied to a wideband audio signal, the listener will only perceive a slight difference, if any. However, if a peak filter with the same bandwidth is applied to the same signal, the listener will perceive a considerable difference.

[0172] The embodiment is based on the finding that band-limited disruption in acoustic contrast results in a peak in acoustic leakage, and this fact can be utilized (see Figure 5). If the acoustic scene reproduced in the bright region is filtered by the notch filter, it will be barely noticeable to listeners in this region. On the other hand, the peak in acoustic leakage perceived in the dark region is compensated for by this measurement.

[0173] An example of the corresponding filter response is shown in Figure 10. In particular, Figure 10 shows an exemplary design of a spectral shaping device according to an embodiment, where (a) shows the acoustic contrast obtained by a specific reproduction method, and (b) shows the selected amplitude response of the spectral shaping filter.

[0174] As outlined above, the filter 140 is configured to generate multiple speaker signals depending on which of the two or more sound regions the two or more sound source signals should be reproduced in, depending on which of the two or more sound regions they should be reproduced in, depending on whether the two or more sound source signals should not be reproduced.

[0175] The following describes a filter 140 according to an embodiment, for example, a pre-filter.

[0176] In one embodiment, for example, one or more sound source signals are reproduced in the first sound region but not in the second sound region, and at least one further sound source signal is reproduced in the second sound region but not in the first sound region.

[0177] TIFF0007844407000068.tif19151

[0178] TIFF0007844407000069.tif38152TIFF0007844407000070.tif32152

[0179] Audio source signal The first Playback occurs in sound domain 1. The first It will not be played in sound domain 2. In order to ,Also teeth , At least, the sound source signal in the first sound domain, Second sound domain In louder than And again to be born In order to (and / or at least, the sound source signal is In the first sound domain, Second sound domain In Larger signal energy And again to be born In order to ) of Appropriate measures can be taken.

[0180] For example, you can use filter 140. The filter coefficients are, For example, a first sound source signal that is reproduced in the first sound domain but not in the second sound domain. However, in the first sound domain , second sound domain In Greater loudness (and / or larger signal) Energy ) And again So that I may be born , selection It can be selected. Furthermore, the filter coefficients are, for example, , the Played in two sound domains. However, it is not reproduced in the first sound domain. Second sound source signal However, in the second sound domain , the first sound domain In Greater loudness (and / or larger signal) Energy ) And again It may be chosen to be born.

[0181] For example, an FIR filter (finite impulse response filter) can be used, and the filter coefficients can be appropriately selected, for example, as described below.

[0182] Alternatively, Wave Field Synthesis (WFS), which is well known in the field of speech processing (for example, as one of many examples

[69] , for general information on Wave Field Synthesis), may be employed.

[0183] Alternatively, Higher-Order Ambis, which is well known in the field of speech processing. You can use onics (for example, Higher-Order Ambiso For general information regarding nics, see one of many examples

[70] .

[0184] Here, we will describe in more detail some specific embodiments of the filter 140.

[0185] A set of multiple loudspeakers is considered a loudspeaker array whenever a router supplies at least one input signal to multiple loudspeakers that are primarily excited in the same frequency range. Each individual loudspeaker is part of multiple arrays, and multiple input signals can be supplied to one array, which then radiate in different directions.

[0186] See [1], [3], [4], [5] and [6], there are well-known different methods for determining a linear prefilter so that an array of omnidirectional loudspeakers exhibits a directional radiation pattern.

[0187] Some embodiments implement pressure matching techniques based on measured impulse responses. Some of these embodiments employing such approaches, described below, consider only a single speaker array. Other embodiments use multiple loudspeaker arrays. Applications to multiple loudspeaker arrays are straightforward.

[0188] TIFF0007844407000072.tif35151

[0189] TIFF0007844407000073.tif64151

[0190] TIFF0007844407000074.tif13151TIFF0007844407000075.tif32145

[0191] TIFF0007844407000076.tif38156

[0192] TIFF0007844407000077.tif101151

[0193] TIFF0007844407000078.tif32151TIFF0007844407000079.tif35152

[0194] TIFF0007844407000080.tif56152

[0195] Maximizing equation (34) can be solved as a generalized eigenvalue problem [3]. We should be mindful of what is possible.

[0196] TIFF0007844407000081.tif40151

[0197] TIFF0007844407000082.tif20146

[0198] TIFF0007844407000083.tif17152TIFF0007844407000084.tif7163

[0199] TIFF0007844407000085.tif13152

[0200] TIFF0007844407000086.tif59148

[0201] TIFF0007844407000087.tif121149TIFF0007844407000088.tif8152

[0202] TIFF0007844407000089.tif15154

[0203] TIFF0007844407000090.tif72154

[0204] Regarding the calculation of the filter coefficients, it is noteworthy that equation (36) explicitly gives the required filter coefficients, making the calculation quite demanding in practice. Due to the similarity between this problem and the listening room equalization problem, the method used there can also be applied.

[0205] Therefore, a very efficient algorithm for calculating equation (36) is found in reference

[71] : SCHNEIDER, Martin; KELLERMANN, Walter: "Iterative DFT-domain inverse filter determination for adaptive listening room equalization." In: Acoustic Signal Enhancement; Proceedings of IWAENC 2012; International Workshop on. VDE, 2012. It is described in sections S. 1-4.

[0206] The following describes a loudspeaker enclosure microphone system (LEMS) according to an embodiment. In particular, the design of the LEMS according to the embodiment will be described. In some embodiments, the above means may depend, for example, on different characteristics of the LEMS.

[0207] Figure 11 shows an exemplary loudspeaker setup in an enclosure according to one embodiment. In particular, Figure 11 shows an exemplary LEMS having four sound regions. Each acoustic scene needs to be reproduced in its respective sound region. For this purpose, the speakers shown in Figure 11 are used in a specific manner in relation to their relative positions to each other and to the sound regions.

[0208] Two speaker arrays, indicated by "Array 1" and "Array 2," are used with a pre-filter determined accordingly (see above). In this manner, it is possible to electrically steer the radiation from these arrays toward "Region 1" and "Region 2." Assuming that both arrays have a speaker-to-speaker distance of a few centimeters and the arrays have aperture sizes of a few decimeters, effective steering is possible with respect to the midrange frequencies.

[0209] Although not entirely clear, for example, omnidirectional speakers "LS1," "LS2," "LS3," and "LS4," which can be positioned 1 to 3 meters apart from each other, are driven as a speaker array, considering frequencies below 300 Hz, for example. This can be determined using the method described above, according to the pre-filter.

[0210] The "LS5" and "LS6" speakers are directional speakers that provide high-frequency sound to regions 3 and 4, respectively.

[0211] As mentioned above, the measure for directional reproduction may not yield satisfactory results across the entire audible frequency range. To compensate for this problem, for example, loudspeakers can be placed nearby or within their respective sound regions. While this arrangement is suboptimal in terms of perceived sound quality, the difference in speaker distance to the assigned region compared to the distance to other regions allows for spatially focused reproduction, regardless of frequency. Therefore, these loudspeakers can be used, for example, in frequency ranges where other methods do not yield satisfactory results.

[0212] TIFF0007844407000091.tif44151

[0213] Since acoustic leakage depends on the playback method selected to differ for each frequency band, such embodiments have the advantage that pre-processing parameters can be adapted to the requirements of the playback method.

[0214] Furthermore, if such an implementation is chosen, compensating for leakage in one frequency band will not affect another frequency band. Since the "preprocessing" block is not an LTI system, this replacement means a change in the functionality of the entire system, even though the entire system will reliably solve the same problem.

[0215] Furthermore, it should be noted that some embodiments may utilize the measurement of impulse responses from all speakers to multiple microphones prior to operation. Therefore, microphones are not required during operation.

[0216] The proposed method is generally suitable for multi-zone simulation scenarios, such as in-vehicle scenarios.

[0217] Depending on the specific implementation requirements, embodiments of the present invention may be implemented in hardware or software, or at least partially in hardware or at least partially in software. Implementations may be carried out using digital storage media such as floppy disks, DVDs, Blu-rays, CDs, ROMs, PROMs, EPROMs, EEPROMs, or flash memory, which store electronically readable control signals, and which cooperate (or can cooperate) with a programmable computer system so that each method is performed. Thus, the digital storage media may be computer-readable.

[0218] Some embodiments of the present invention include a data carrier having an electronically readable control signal, which, in cooperation with a programmable computer system, enables one of the methods described herein to be performed.

[0219] Generally, embodiments of the present invention can be implemented as a computer program product having program code that operates to perform one of the methods when the computer program product runs on a computer. The program code can be stored, for example, on a machine-readable carrier.

[0220] Other embodiments include a computer program stored in a machine-readable carrier for performing one of the methods described herein.

[0221] In other words, an embodiment of the method of the present invention is a computer program having program code for performing one of the methods of the present invention when the computer program is executed on a computer.

[0222] Accordingly, a further embodiment of the method of the present invention is a data carrier (or digital storage medium or computer-readable medium) containing a computer program for performing one of the methods described herein. The data carrier, digital storage medium or recording medium is typically tangible and / or non-temporary.

[0223] Therefore, a further embodiment of the method of the present invention is a data stream or sequence of signals representing a computer program for performing one of the methods described herein. The data stream or sequence of signals is transmitted, for example, via the Internet. It can be configured to be transferred via a data communication connection.

[0224] Further embodiments include processing means configured to perform or applied to perform one of the methods described herein, such as a computer or a programmable logic device.

[0225] Further embodiments include a computer on which a computer program for performing one of the methods described herein is installed.

[0226] Further embodiments of the present invention include an apparatus or system configured to transfer (e.g., electronically or optically) a computer program for performing one of the methods described herein to a receiver. The receiver may be, for example, a computer, a mobile device, a memory device, etc. The apparatus or system may include, for example, a file server for transferring the computer program to the receiver.

[0227] In some embodiments, a programmable logic device (e.g., a field-programmable gate array) can be used to perform some or all of the functions of the methods described herein. In some embodiments, a field-programmable gate array can cooperate with a microprocessor to perform one of the methods described herein. Generally, these methods are preferably performed by any hardware device.

[0228] The apparatus described herein can be implemented using hardware devices, a computer, or a combination of hardware devices and a computer.

[0229] The methods described herein may be performed using hardware devices, or using a computer, or using a combination of hardware devices and a computer.

[0230] The embodiments described above are merely illustrative of the principles of the present invention. Modifications and variations of the configurations and details described herein will be apparent to those skilled in the art. Therefore, it is intended that the invention is limited only by the immediate claims and not by the description of the embodiments herein and the specific details shown herein.

[0231] References [1] W. Druyvesteyn and J. Garas, "Personal sound," Journal of the Audio Engineering Society, vol. 45, no. 9, pp. 685-701, 1997. [2] F. Dowla and A. Spiridon, “Spotforming with an array of ultra-wideband radio transmitters," in Ultra Wideband Systems and Technologies, 2003 IEEE Conference on, Nov 2003, pp. 172-175. [3] J.-W. Choi and Y.-H. Kim, "Generation of an acoustically bright zone with an illuminated region using multiple sources," Journal of the Acoustical Society of America, vol. 111, no. 4, pp. 1695-1700, 2002. [4] M. Poletti, "An investigation of 2-d multizone surround sound systems," in Audio Engineering Society Convention 125, Oct 2008. [Online]. Available: http: / / www.aes.org / e-lib / browse.cfm-elib=14703 . [5] Y. Wu and T. Abhayapala, "Spatial multizone soundfield reproduction," in Acoustics, Speech and Signal Processing, 2009. ICASSP 2009. IEEE International Conference on, April 2009, pp. 93-96. [6] Y. J. Wu and T. D. Abhayapala, "Spatial multizone soundfield reproduct ion: Theory and design," Audio, Speech, and Language Processing, IEEE Transactions on, vol. 19, no. 6, pp. 1711-1720, 2011. [7] D. Brandwood, "A complex gradient operator and its application in adaptive array theory," Microwaves, Optics and Antennas, IEE Proceedings H, vol. 130, no. 1, pp. 11 - 16, Feb. 1983. [8] U.S. Patent Application Publication No. 2005 / 0152562 [9] U.S. Patent Application Publication No. 2013 / 170668

[10] U.S. Patent Application Publication No. 2008 / 0071400

[11] U.S. Patent Application Publication No. 2006 / 0034470

[12] U.S. Patent Application Publication No. 2011 / 0222695

[13] U.S. Patent Application Publication No. 2009 / 0232320

[14] U.S. Patent Application Publication No. 2015 / 0256933

[15] U.S. Patent No. 6,674,865

[16] German Patent Application Publication No. 3045722

[17] U.S. Patent Application Publication No. 2012 / 0140945

[18] U.S. Patent Application Publication No. 2008 / 0273713

[19] U.S. Patent Application Publication No. 2004 / 0105550

[20] U.S. Patent Application Publication No. 2006 / 0262935

[21] U.S. Patent Application Publication No. 2005 / 019035

[22] U.S. Patent Application Publication No. 2008 / 0130922

[23] U.S. Patent Application Publication No. 2010 / 0329488

[24] German Patent Application Publication No. 102014210105

[25] U.S. Patent Application Publication No. 2011 / 0286614

[26] U.S. Patent Application Publication No. 2007 / 0053532

[27] U.S. Patent Application Publication No. 2013 / 0230175

[28] International Publication No. 2016 / 008621

[29] U.S. Patent Application Publication No. 2008 / 0273712

[30] U.S. Patent No. 5,870,484

[31] U.S. Patent No. 5,809,153

[32] U.S. Patent Application Publication No. 2006 / 0034467

[33] U.S. Patent Application Publication No. 2003 / 0103636

[34] U.S. Patent Application Publication No. 2003 / 0142842

[35] Japanese Patent No. 5345549

[36] U.S. Patent Application Publication No. 2014 / 0056431

[37] U.S. Patent Application Publication No. 2014 / 0064526

[38] U.S. Patent Application Publication No. 2005 / 0069148

[39] U.S. Patent No. 5,081,682

[40] Specification of German Utility Model Registration No. 9015454

[41] U.S. Patent No. 5,550,922

[42] U.S. Patent No. 5,434,922

[43] U.S. Patent No. 6,078,670

[44] U.S. Patent No. 6,674,865

[45] German Patent Application Publication No. 10052104

[46] U.S. Patent Application Publication No. 2005 / 0135635

[47] German Patent Application Publication No. 10242558

[48] ​​U.S. Patent Application Publication No. 2010 / 0046765

[49] German Patent Application Publication No. 102010040689

[50] U.S. Patent Application Publication No. 2008 / 0103615

[51] U.S. Patent No. 8,190,438B1

[52] International Publication No. 2007 / 098916

[53] U.S. Patent Application Publication No. 2007 / 0274546

[54] U.S. Patent Application Publication No. 2007 / 0286426

[55] U.S. Patent No. 5,018,205

[56] U.S. Patent No. 4,944,018

[57] German Patent Application Publication No. 10351145

[58] Japanese Patent Publication No. 2003-255954

[59] U.S. Patent No. 4,977,600

[60] U.S. Patent No. 5,416,846

[61] U.S. Patent Application Publication No. 2007 / 0030976

[62] Japanese Patent Application Publication No. 2004-363696

[63] Wikipedia: "Angular resolution", https: / / en.wikipedia.org / wiki / Angular_resolution , retrieved from the Internet on 8 April 2016.

[64] Wikipedia: "Nyquist-Shannon sampling theorem", https: / / en.wikipedia.org / wiki / Nyquist-Shannon_sampling_theorem, retrieved from the Internet on 8 April 2016.

[65] Wikipedia: "Dynamic range compression", https: / / en.wikipedia.org / wiki / Dynamic_range_compression, retrieved from the Internet on 8 April 2016.

[66] Wikipedia: "Weighting filter", https: / / en.wikipedia.org / wiki / Weighting_filter , retrieved from the Internet on 8 April 2016.

[67] Wikipedia: "Audio crossover - Digital" , https: / / en.wikipedia.org / wiki / Audio_crossover#Digital , retrieved from the Internet on 8 April 2016.

[68] Wikipedia: "Equalization (audio) - Filter functions", https: / / en.wikipedia.org / wiki / Equalization_(audio)_Filter_functions , retrieved from the Internet on 8 April 2016.

[69] International Publication No. 2004 / 114725

[70] European Patent Application Publication No. 2450880

[71] SCHNEIDER, Martin; KELLERMANN, Walter: "Iterative DFT-domain inverse filter determination for adaptive listening room equalization." In: Acoustic Signal Enhancement; Proceedings of IWAENC 2012; International Workshop on. VDE, 2012, S. 1-4.

Claims

1. A device for generating multiple speaker signals from two or more sound source signals, wherein each of the two or more sound source signals is reproduced in one or more of two or more sound regions, and at least one of the two or more sound source signals is not reproduced in at least one of the two or more sound regions, and the device is A speech preprocessing device (110) configured to modify each of two or more initial speech signals to obtain two or more preprocessed speech signals, A filter (140) configured to generate the plurality of speaker signals depending on the two or more preprocessed audio signals, Includes, The audio preprocessor (110) is configured to use the two or more sound source signals as the two or more initial audio signals, or the audio preprocessor (110) is configured to generate one of the two or more initial audio signals by modifying each of the two or more sound source signals. The audio preprocessor (110) is configured to modify each of the two or more initial audio signals depending on the signal power or loudness of another initial audio signal among the two or more initial audio signals. The filter (140) is configured to generate the plurality of speaker signals using filter coefficients selected such that the first sound source signal among the two or more sound source signals, which is reproduced in the first sound region but not in the second sound region, is reproduced in the first sound region with greater loudness and / or greater signal energy than in the second sound region, and the second signal among the two or more sound source signals, which is reproduced in the second sound region but not in the first sound region, is reproduced in the second sound region with greater loudness and / or greater signal energy than in the first sound region. The audio preprocessor (110) is configured to modify each of the two or more initial audio signals in accordance with the ratio of a first value to a second value, thereby modifying each initial audio signal among the two or more initial audio signals in accordance with the signal power or loudness of another initial audio signal among the two or more initial audio signals. The second value depends on the signal power of the initial audio signal, and the first value depends on the signal power of the other initial audio signal among the two or more initial audio signals, or The second value depends on the loudness of the initial audio signal, and the first value depends on the loudness of the other initial audio signal among the two or more initial audio signals. Device.

2. The audio preprocessor (110) is configured to modify each of the two or more initial audio signals depending on the signal power or loudness of another initial audio signal among the two or more initial audio signals by determining the gain for the initial audio signal and applying the gain to the initial audio signal. The apparatus according to claim 1, wherein the audio preprocessor (110) is configured to determine the gain depending on the ratio between the first value and the second value, the ratio being the ratio between the signal power of the other initial audio signal among the two or more initial audio signals and the signal power of the initial audio signal as the second value, or the ratio being the ratio between the loudness of the other initial audio signal among the two or more initial audio signals and the loudness of the initial audio signal as the second value.

3. The apparatus according to claim 2, wherein the audio preprocessing device (110) is configured to determine the gain depending on a function that increases monotonically with respect to the ratio of the first value to the second value.

4.

5.

6.

7. The apparatus according to any one of claims 1 to 6, wherein the audio preprocessing device (110) is configured to generate the two or more initial audio signals by normalizing the power of each of the two or more sound source signals.

8.

9.

10. The apparatus according to any one of claims 1 to 9, wherein the filter (140) is configured to generate the plurality of speaker signals depending on which of the two or more sound regions the two or more sound source signals are reproduced in, and depending on which of the two or more sound regions the two or more sound source signals are not reproduced in.

11.

12. The apparatus according to any one of claims 1 to 9, wherein the filter (140) is configured to generate the plurality of speaker signals depending on which of the two or more sound regions the two or more sound source signals are reproduced in, and depending on which of the two or more sound regions the two or more sound source signals are not reproduced in.

13. The apparatus further includes two or more band dividers (121, 122) configured to perform band division of the two or more preprocessed audio signals into a plurality of band-divided audio signals, The filter (140) is configured to generate the plurality of speaker signals depending on the plurality of band-divided audio signals. The apparatus according to any one of claims 1 to 12.

14. The apparatus further includes one or more spectral shapers (131, 132, 133, 134) configured to modify the spectral envelope of one or more of the plurality of band-divided audio signals to obtain one or more spectrally shaped audio signals, The filter (140) is configured to generate the plurality of speaker signals depending on one or more spectrally shaped audio signals. The apparatus according to claim 13.

15. A method for generating multiple speaker signals from two or more sound source signals, wherein each of the two or more sound source signals is reproduced in one or more of two or more sound regions, and at least one of the two or more sound source signals is not reproduced in at least one of the two or more sound regions, and the method is The steps include modifying each of two or more initial audio signals to obtain two or more preprocessed audio signals, A step of generating the plurality of speaker signals depending on the two or more preprocessed audio signals, Includes, The two or more sound source signals are used as the two or more initial audio signals, or, for each of the two or more sound source signals, one of the two or more initial audio signals is generated by modifying the sound source signal. Each of the two or more initial audio signals is modified depending on the signal power or loudness of another initial audio signal among the two or more initial audio signals. The plurality of speaker signals are generated using a filter with filter coefficients selected such that the first sound source signal among the two or more sound source signals, which is reproduced in the first sound region but not in the second sound region, is reproduced in the first sound region with greater loudness and / or greater signal energy than in the second sound region, and the second sound source signal among the two or more sound source signals, which is reproduced in the second sound region but not in the first sound region, is reproduced in the second sound region with greater loudness and / or greater signal energy than in the first sound region. The method includes the step of modifying each of the two or more initial audio signals in accordance with the ratio of a first value to a second value, thereby modifying each initial audio signal among the two or more initial audio signals in accordance with the signal power or loudness of another initial audio signal among the two or more initial audio signals. The second value depends on the signal power of the initial audio signal, and the first value depends on the signal power of the other initial audio signal among the two or more initial audio signals, or The second value depends on the loudness of the initial audio signal, and the first value depends on the loudness of the other initial audio signal among the two or more initial audio signals. method.

16. A computer program for carrying out the method according to claim 15 when executed on a computer or signal processor.

Images