EP3069530B1 - Method and device for compressing and decompressing sound field data of an area - Google Patents

Description

The present invention relates to audio technology, and more particularly to the compression of spatial sound field data.

The acoustic description of rooms is of great interest for the control of display devices in the form of e.g. a headphone, a speaker assembly with e.g. two to a mean number of loudspeakers, such as 10 loudspeakers or loudspeaker arrangements with a large number of loudspeakers, as used in Wave Field Synthesis (WFS).

There are various approaches to spatial audio coding in general. One approach is e.g. in creating different channels for different speakers at predefined speaker positions, as is the case with MPEG surround, for example. As a result, a listener located in the reproduction room at a certain and, ideally, the middle position, obtains a sense of space for the reproduced sound field.

An alternative spatial description is to describe a space through its impulse response. For example, if a sound source is positioned anywhere in a room or area, that space or area may be measured with a circular array of microphones in the case of a two-dimensional area or with a ball-microphone array in the case of a three-dimensional area. For example, if a ball-microphone array with a high number of microphones is considered, such as 350 microphones, then a survey of the room will proceed as follows. At a certain position inside or outside the microphone array, a pulse is generated. Then the response to this pulse, ie the impulse response, is measured by each microphone. Depending on how strong the reverb characteristics are, a longer or shorter impulse response is then measured. For example, in large scale measurements in large churches have shown that impulse responses can last over 10 s.

Such a set of e.g. 350 impulse responses describe the sound characteristics of this room for the specific position of a sound source where the impulse was generated. In other words, this set of impulse responses represents sound field data of the area for exactly the one case where a source is positioned at the position where the impulse was generated. In order to further measure the space, so as to detect the sound properties of the room when a source is positioned at another room, the illustrated procedure must be repeated for each additional position e.g. be repeated outside the array (but also inside the array). Therefore, if, for example, one were to record a concert hall by sound field, if e.g. playing a musician quartet in which the individual musicians are arranged in four different positions, so 350 impulse responses are measured for each of the four positions in the example mentioned, and these 4 x 350 = 1400 impulse responses then represent the sound field data of the area.

Since the temporal length of the impulse responses can be quite considerable, and since a more detailed representation of the sound properties of the room may be desired in terms of not only four but more positions, there is a huge amount of impulse response data, especially when considered is that the impulse responses can well take lengths over 10 s.

Approaches for spatial audio coding is e.g. spatial audio coding (SAC) [1] or spatial audio object coding (SAOC) [2], which enable a bit rate-efficient coding of multi-channel audio signals or object-based spatial audio scenes. Spatial impulse response rendering (SIRR) [3] and directional audio coding (DirAc) [4] are parametric coding techniques based on a time-dependent direction of arrival (DOA) estimation and an estimate of the diffusivity within frequency bands. Here, a separation between non-diffuse and diffuse sound field is made. [5] deals with the lossless compression of spherical microphone array data and the coding of higher order Ambisonics signals. Compression is achieved by exploiting redundant data between the channels (interchannel redundancy).

1. [1] Herre, J et al (2004) Spatial Audio Coding: Next-generation efficient and compatible coding of multi-channel audio AES Convention Paper 6186 presented at the 117th Convention, San Francisco, USA
2. [2] Engdegard, J et al (2008) Spatial Audio Object Coding (SAOC) - The Upcoming MPEG Standard on Parametric Object Based Audio Coding, AES Convention Paper 7377 presented at the 125th Convention, Amsterdam, Netherlands
3. [3] Merimaa J and Pulkki V (2003) Perceptually-based processing of directional room responses for multichannel loudspeaker reproduction, IEEE Workshop on Applications of Signal Processing to Audio and Acoustics
4. [4] Pulkki, V (2007) Spatial Sound Reproduction with Directional Audio Coding, J. Audio Eng. Soc., Vol. 55. No.6
5. [5] Hellerud E et al (2008) Encoding Higher Order Ambisonics with AAC AES Convention Paper 7366 presented at the 125th Convention, Amsterdam, Netherlands
6. [6] Liindau A, Kosanke L, Weinzierl S (2010) Perceptual evaluation of physical predictors of the mixing time in binaural room impulse responses AES Convention Paper presented at the 128th Convention, London, UK
7. [7] Avni, A and Rafaely B (2009) Interaural cross correlation and spatial correlation in a sound field represented by spherical harmonics in Ambisonics Symposium 2009, Graz, Austria

Investigations in [6] show a separate consideration of early and late sound field in binaural reproduction. For dynamic systems, where head movements are considered, the filter length is optimized by using only the early sound field in Real time is folded. For the late sound field, only one filter is sufficient for all directions without reducing the perceived quality. In [7] head-related transfer functions (HRTF) are represented on a sphere in the spherical harmonic range. The influence of different accuracies by means of different orders of spherical harmonics on the interaural cross-correlation and the spatio-temporal correlation is investigated analytically. This happens in octave bands in the diffuse sound field.

1. [1] Herre, J et al (2004) Spatial Audio Coding: Next-generation efficient and compatible coding of multi-channel audio AES Convention Paper 6186 presented at the 117th Convention, San Francisco, USA
2. [2] Engdegard, J et al (2008) Spatial Audio Object Coding (SAOC) - The Upcoming MPEG Standard on Parametric Object Based Audio Coding, AES Convention Paper 7377 presented at the 125th Convention, Amsterdam, Netherlands
3. [3] Merimaa J and Pulkki V (2003). Perceptually-based processing of directional room responses for multichannel loudspeaker reproduction, IEEE Workshop on Signal Processing to Audio and Acoustics
4. [4] Pulkki, V (2007) Spatial Sound Reproduction with Directional Audio Coding, J. Audio Eng. Soc., Vol. 55. No.6
5. [5] Hellerud E et al (2008) Encoding Higher Order Ambisonics with AAC AES Convention Paper 7366 presented at the 125th Convention, Amsterdam, Netherlands
6. [6] Liindau A, Kosanke L, Weinzierl S (2010) Binaural room impulse responses AES Convention Paper presented at the 128th Convention, London, UK
7. [7] Avni, A and Rafaely B (2009) Interaural cross correlation and spatial correlation in a field of sound represented by spherical harmonics at Ambisonics Symposium 2009, Graz, Austria

• [10] beschreibt ein parametrisches, kanalbasiertes Audiokodierverfahren im Zeit- und Frequenzbereich. In [11] wird ein binaural-cue-coding (BCC) beschrieben, das ein oder mehr objektbasierte Cue-Codes verwendet. Diese beinhalten Richtung, Weite und Umhüllung einer auditorischen Szene. [12] bezieht sich auf die Verarbeitung von Kugelarraydaten für die Wiedergabe mittels Ambisonics. Dabei sollen die Verzerrungen des Systems durch Messfehler, wie z.B. Rauschen, equalisiert werden. In [13] wird ein kanalbasiertes Kodierverfahren beschrieben, dass sich auch auf Positionen der Lautsprecher, sowie einzelner Audio Objekte bezieht. In [14] wird ein Matrix-basiertes Kodierverfahren vorgestellt, das die Echtzeitübertragung von Higher Order Ambisonics Schallfeldern mit Ordnungen größer als 3 ermöglicht.
  In [15] wird eine Methode zur Kodierung von räumlichen Audiodaten beschrieben, das unabhängig vom Wiedergabesystem ist. Dabei wird das Eingangsmaterial in zwei Gruppen unterteilt, von denen die erste Gruppe das Audio beinhaltet, das hohe Lokalisierbarkeit benötigt, während die zweite Gruppe mit für die Lokalisation ausreichend niedrigen Ambisonics-Ordnungen beschrieben wird. In der ersten Gruppe wird das Signal in einen Satz aus Monokanälen mit Metadaten kodiert. Die Metadaten beinhalten Zeitinformationen, wann der entsprechende Kanal wiedergegeben werden soll und Richtungsinformationen zu jedem Moment. Bei der Wiedergabe werden die Audiokanäle für herkömmliche Panning-Algorithmen dekodiert, wobei das Wiedergabe-System bekannt sein muss. Das Audio in der zweiten Gruppe wird in Kanäle verschiedener Ambisonics-Ordnungen kodiert. Bei der Dekodierung werden dem Wiedergabesystem entsprechende Ambisonics-Ordnungen verwendet.
• [8] Dolby R M (1999) Low-bit-rate spatial coding method and system, EP 1677576 A3
• [9] Goodwin M and Jot J-M (2007) Spatial audio coding based on universal spatial cues, US 8,379,868 B2
• [10] Seefeldt A and Vinton M (2006) Controlling spatial audio coding parameters as a function of auditory events, EP 2296142 A2
• [11] Faller C (2005) Parametric coding of spatial audio with object-based side information, US 8340306 B2
• [12] Kordon S, Batke J-M, Krüger A (2011) Method and apparatus for processing signals of a spherical microphone array on a rigid sphere used for generating an ambisonics representation of the sound field, EP 2592845 A1
• [13] Corteel E and Rosenthal M (2011) Method and device for enhanced sound field reproduction of spatially encoded audio input signals, EP 2609759 A1
• [14] Abeling S et al (2010) Method and apparatus for generating and for decoding sound field data including ambisonics sound field data of an order higher than three, EP 2451196 A1
• [15] Arumi P and Sole A (2008) Method and apparatus for three-dimensional acoustic field encoding and optimal reconstruction, EP 2205007 A1

An encoder-decoder scheme for low bit rates is described in [8]. The encoder generates a composite audio information signal representing the sound field to be reproduced describes and a directional vector or steering control signal. The spectrum is divided into subbands. For control, the dominant direction is evaluated in each subband. Based on the perceived spatial audio scene, [9] describes a spatial audio coding framework in the frequency domain. Time-frequency dependent direction vectors describe the input audio scene.

• [10] describes a parametric, channel-based audio coding method in the time and frequency domain. [11] describes a binaural-cue-coding (BCC) that uses one or more object-based cue codes. These include direction, width and envelopment of an auditory scene. [12] refers to the processing of sphere array data for rendering by Ambisonics. The distortion of the system should be equalized by measurement errors, such as noise. In [13] a channel-based coding method is described, which also refers to positions of the loudspeakers, as well as individual audio objects. In [14] a matrix-based coding method is presented, which allows the real-time transmission of Higher Order Ambisonics sound fields with orders greater than 3.
  [15] describes a method for encoding spatial audio that is independent of the rendering system. The input material is subdivided into two groups, the first of which contains the audio, which requires high localizability, while the second group is described with ambisonics orders that are sufficiently low for localization. In the first group, the signal is encoded into a set of mono channels with metadata. The metadata includes time information as to when the corresponding channel is to be played back and directional information at each moment. During playback, the audio channels are decoded for conventional panning algorithms, where the playback system must be known. The audio in the second group is encoded into channels of different Ambisonics orders. During decoding, the Ambiusics orders are used in the playback system.
• [8] Dolby RM (1999) Low-bit-rate spatial coding method and system, EP 1677576 A3
• [9] Goodwin M and Jot JM (2007) Spatial audio coding based on universal spatial cues, US 8,379,868 B2
• [10] Seefeldt A and Vinton M (2006) Controlling spatial audio coding parameters as a function of auditory events, EP 2296142 A2
• [11] Faller C (2005) Parametric coding of spatial audio with object-based side information US 8340306 B2
• [12] Kordon S, Batke JM, Kruger A (2011) Method and apparatus for processing signals of a spherical microphone array on a rigid sphere used for generating an ambisonics representation of the sound field, EP 2592845 A1
• [13] Corteel E and Rosenthal M (2011) Method and device for enhanced sound field reproduction of spatially encoded audio input signals, EP 2609759 A1
• [14] Abeling S et al. (2010) Method and apparatus for generating and decoding sound field data including ambisonics. EP 2451196 A1
• [15] Arumi P and Sole A (2008) Method and apparatus for three-dimensional acoustic field encoding and optimal reconstruction, EP 2205007 A1

The specialist publication " Development and evaluation of a mixed-order Ambisonics playback system ", J. Käsbach, Technical University of Denmark, November 2010 discloses a mixed-order ambisonic playback system in which the decomposition into spherical harmonic components of the three-dimensional sound field has been supplemented by additional horizontal components. Considering the orthonormality properties of the spherical functions, the maximum two-dimensional and three-dimensional orders are determined for a given speaker array. Based on this analysis, an alternative mixed-order implementation is proposed which requires a truncated order of inherent Legendre functions.

The specialist publication " A New Mixed-Order Scheme for Ambisonic Signals ", Chris Travis, Ambisonics Symposium 2009, June 25, 2009, Graz, pp. 1-6 refers to mixed-order systems that provide higher resolution in the horizontal plane than at the poles. A two-parameter scheme (# H # V) intersects the spherical harmonic Extension in a certain way. It results in resolution-versus-elevation curves that are flatter at and near the horizontal plane.

The WO 2010/012478 A2 discloses a system for generating binaural signals based on a multi-channel signal representing a plurality of channels and intended for reproduction by a speaker configuration having an associated virtual sound source position for each channel.

The specialist publication " Multichannel Audio Coding Based on Analysis by Synthesis ", I. Elfitri, et al., Proceedings of the IEEE, New York, vol. 99, no. 4, 1.4.2011, pages 657-670 describes a closed-loop encoding system based on an analysis-by-synthesis principle applied to the MPEG surround architecture.

The object of the present invention is to provide a more efficient concept for handling such. B. compressing or decompressing sound field data of a region.

This object is achieved by a sound field data compression device according to claim 1, a sound field data decompression device according to claim 13, a sound field data compression method according to claim 19, a sound field data decompression method according to claim 20 or a computer program according to claim 21 ,

A device for compressing sound field data of a region comprises a splitter for splitting the sound field data into a first part and a second part, and a downstream converter for converting the first part and the second part into harmonic components, wherein the conversion takes place so that the second Number is converted into one or more harmonic components with a second order, and that the first portion in harmonic components with ei first order, with the first order being higher than the second order, to obtain the compressed sound field data.

Thus, an implementation of the sound field data, such as the amount of impulse responses is performed in harmonic components, already this implementation can lead to significant data savings. Harmonic components, such as are obtainable by means of a spatial spectral transformation, describe a sound field much more compact than impulse responses. In addition, the order of harmonic components is readily controllable. The harmonic component of zeroth order is just a (non-directional) mono signal. It does not yet allow a sound field direction description. By contrast, the additional first order harmonic components already allow a relatively coarse directional representation analogous to beamforming. The harmonic components of second order allow an even more accurate sound field description with even more directional information. For example, in Ambisonics, the number of components is 2n + 1, where n is the order. For the zeroth order there is thus only a single harmonic component. There are already three harmonic components for implementation up to the first order. For example, for a fifth-order conversion, there are already 11 harmonic components, and it has been found that, for example, for 350 impulse responses, an order equal to 14 is sufficient. In other words, this means that 29 harmonic components describe space as well as 350 impulse responses. Already this conversion from a value of 350 input channels to 29 output channels brings a compression gain. Moreover, implementation of various portions of the sound field data, such as the impulse responses with different orders, is still performed because it has been found that not all proportions need to be described with the same accuracy / order.
An example of this is that the directional perception of the human ear is mainly derived from the early reflections, while the late / diffuse reflections in a typical impulse response to directional perception contribute little or nothing. Thus, in this example, the first portion will be the early portion of the impulse responses, which will be translated to the higher order in the harmonic component range, while the late diffused portion will be converted to a lower order and sometimes even zero order.

Another example is that the directional perception of human hearing is frequency-dependent. At low frequencies, the directional perception of the human ear is relatively weak. For the compression of sound field data, it is therefore sufficient to convert the low spectral range of the harmonic components with a relatively small order in the harmonic component range, while the frequency ranges of the sound field data in which the direction perception of the human ear is very high, with a high and preferably even be implemented with the maximum order. For this purpose, according to the invention, the sound field data is decomposed by means of a filter bank into individual subband sound field data and these subband sound field data are then decomposed with different orders, again the first portion has subband sound field data at higher frequencies, while the second portion has subband sound field data at lower frequencies , where very low frequencies can again be represented even with an order equal to zero, ie only with a single harmonic component.
In another example, the advantageous characteristics of temporal and frequency processing are combined. Thus, the early portion, which is anyway implemented with a higher order, can be decomposed into spectral components, for which then again orders adapted to the individual bands can be obtained. In particular, when a decimating filter bank, such as a QMF filter bank (QMF = Quadrature Mirror Filterbank) is used for the subband signals, the effort to convert the subband sound field data in the harmonic component area is additionally reduced. Moreover, the differentiation of different parts of the sound field data with respect to the order to be computed provides a considerable reduction of the computational effort, since the calculation of the harmonic components, such as the cylindrical harmonic components or the spherical harmonic components strongly depends on the order up to which the harmonics Components are to be calculated. A calculation of the harmonic components to the second order, for example, requires much less computational effort and thus computation time or battery performance, especially in mobile devices as a calculation of the harmonic components to order 14, for example.
In the exemplary embodiments described, the converter is thus designed to implement the component, that is to say the first component of the sound field data, which is more important for a directional perception of the human ear, with a higher order than the second part, which is less important for the directional perception of a sound source than the first part.

The present invention can be used not only for a temporal decomposition of the sound field data into shares or for a spectral decomposition of the sound field data into shares, but also for an alternative, for. B. spatial decomposition of the shares, for example, if it is considered that the direction perception of the human ear for sound in different azimuth or elevation angles is different. For example, if the sound field data is impulse responses or other sound field descriptions where each description is assigned a particular azimuth / elevation angle, the sound field data may be translated from azimuth / elevation angles, where directional perception of the human ear is stronger Order are compressed as a spatial portion of the sound field data from another direction.

Alternatively or additionally, the individual harmonics can be "thinned out", ie in the example with order 14, in which there are 29 modes. Depending on the human sense of direction, individual modes are saved which image the sound field for unimportant sound incidence directions. In the case of microphone array measurements, there is an uncertainty because one does not know in which direction the head is aligned with respect to the array sphere. But if you represent HRTFs by means of spherical harmonics, this uncertainty is resolved.

Further decompositions of the sound field data in addition to decompositions in temporal, spectral or spatial direction can also be used, such as a decomposition of the sound field data into a first and a second portion in volume classes, etc.

In embodiments, the description of acoustic problems occurs in the cylindrical or spherical coordinate system, ie by means of complete sets of orthonormal eigenfunctions, the so-called cylindrical or spherical harmonic components. With higher spatial accuracy of the description of the sound field increase the amount of data and the computing time in the processing or manipulation of the data. For high-quality audio applications, high accuracies are required, which adds to the problems of long computation times, which are particularly detrimental to real-time systems, of large data volumes, which makes transmission more spatial Sound field data difficult, and high energy consumption by intensive computational effort, especially in mobile devices leads.

All of these disadvantages are alleviated or eliminated by embodiments of the invention because, due to the differentiation of the orders for calculating the harmonic components, the calculation times are reduced compared to a case in which all the highest order components are harmonic components be implemented. The large amounts of data are reduced according to the invention in such a way that the representation by harmonic components is already more compact and that additionally different proportions are represented with different orders, wherein the data volume reduction is achieved in that a low order, such as the first order only three has harmonic components, while the highest order has, for example, 29 harmonic components, using the example of an order of 14.

The reduced computing power and the reduced storage volume automatically reduce the energy consumption, which is particularly incurred when using sound field data in mobile devices.

In embodiments, the spatial sound field description is optimized in the cylindrical or spherical harmonic region based on the spatial perception of humans. In particular, a combination of time- and frequency-dependent calculation of the order of spherical harmonics as a function of the spatial perception of human hearing leads to a considerable effort reduction without reducing the subjective quality of sound field perception. Of course, the objective quality is reduced since the present invention represents lossy compression. However, this lossy compression is not critical, especially since the ultimate receiver is the human ear, and therefore it is irrelevant even for transparent reproduction, whether or not sound field components, which are not perceived by the human ear anyway, are present in the reproduced sound field.

In other words, therefore, in the reproduction / auralization either binaurally, ie with headphones or with loudspeaker systems with few (eg stereo) or many speakers (eg WFS) human hearing is the most important quality measure. According to the accuracy of harmonic components such as the cylindrical or spherical harmonic in the time domain and / or in the frequency range and / or in other areas gehörangepasst reduced. This achieves the data and computing time reduction.

Fig. 1a

ein Blockdiagramm einer Vorrichtung zum Komprimieren von Schallfelddaten gemäß einem Ausführungsbeispiel;

Fig. 1b

ein Blockdiagramm einer Vorrichtung zum Dekomprimieren von komprimierten Schallfelddaten eines Gebiets;

Fig. 1c

ein Blockdiagramm einer Vorrichtung zum Komprimieren mit zeitlicher Zerlegung;

Fig. 1d

ein Blockdiagramm eines Ausführungsbeispiels einer Vorrichtung zum Dekomprimieren für den Fall einer zeitlichen Zerlegung;

Fig. 1e

eine zu Fig. 1d alternative Vorrichtung zum Dekomprimieren;

Fig. 1f

ein Beispiel für die Anwendung der Erfindung mit zeitlicher und spektraler Zerlegung am Beispiel von 350 gemessenen Impulsantworten als Schallfelddaten;

Fig. 2a

ein Blockdiagramm einer Vorrichtung zum Komprimieren mit spektraler Zerlegung;

Fig. 2b

ein Beispiel einer unterabgetasteten Filterbank und einer anschließenden Umsetzung der unterabgetasteten Subband-Schallfelddaten;

Fig. 2c

eine Vorrichtung zum Dekomprimieren für das in Fig. 2a gezeigte Beispiel der spektralen Zerlegung;

Fig. 2d

eine alternative Implementierung des Dekomprimierers für die spektrale Zerlegung;

Fig. 3a

ein Übersichts-Blockdiagramm mit einem speziellen Analyse/SyntheseCodierer gemäß einem weiteren Ausführungsbeispiel der vorliegenden Erfindung;

Fig. 3b

eine detailliertere Darstellung eines Ausführungsbeispiels mit zeitlicher und spektraler Zerlegung;

Fig. 4

eine schematische Darstellung einer Impulsantwort;

Fig. 5

ein Blockschaltbild eines Umsetzers vom Zeit- oder Spektralbereich in den Harmonischen-Komponenten-Bereich mit variabler Ordnung; und

Fig. 6

eine Darstellung eines beispielhaften Umsetzers vom HarmonischenKomponenten-Bereich in den Zeitbereich oder Spektralbereich mit anschließender Auralisation.

Preferred embodiments of the present invention will be explained below in detail with reference to the accompanying drawings. Show it:

Fig. 1a

a block diagram of a device for compressing sound field data according to an embodiment;

Fig. 1b

a block diagram of an apparatus for decompressing compressed sound field data of a region;

Fig. 1c

a block diagram of a device for compression with temporal decomposition;

Fig. 1d

a block diagram of an embodiment of a device for decompressing in the case of a temporal decomposition;

Fig. 1e

one too Fig. 1d alternative device for decompressing;

Fig. 1f

an example of the application of the invention with temporal and spectral decomposition on the example of 350 measured impulse responses as sound field data;

Fig. 2a

a block diagram of a device for compressing with spectral decomposition;

Fig. 2b

an example of a sub-sampled filter bank and a subsequent conversion of the subsampled subband sound field data;

Fig. 2c

a device for decompressing the in Fig. 2a shown example of the spectral decomposition;

Fig. 2d

an alternative implementation of the decompressor for spectral decomposition;

Fig. 3a

an overview block diagram with a special analysis / synthesis encoder according to another embodiment of the present invention;

Fig. 3b

a more detailed representation of an embodiment with temporal and spectral decomposition;

Fig. 4

a schematic representation of an impulse response;

Fig. 5

a block diagram of a converter from the time or spectral range in the harmonic component variable-order area; and

Fig. 6

a representation of an exemplary converter from the harmonic component area in the time domain or spectral range with subsequent auralization.

Fig. 1a FIG. 12 shows a block diagram of a device or method for compressing sound field data of a region as input at an input 10 into a splitter 100. The splitter 100 is designed to divide the sound field data into a first portion 101 and a second portion 102. In addition, a converter is provided which has the two functionalities designated 140 or 180. In particular, the converter is configured to convert the first portion 101, as shown at 140, and to convert the second portion 102, as shown at 180. In particular, the converter converts the first portion 101 into one or more harmonic components 141 having a first order, while the converter 180 converts the second portion 102 into one or more harmonic components 182 having a second order. In particular, the first order, that is, the order underlying the harmonic components 141 is higher than the second order, which means, in other words, that the higher order converter 140 outputs more harmonic components 141 than the lower order converter 180. The order n ₁ , by which the converter 140 is driven, is thus greater than the order n ₂ , with which the converter 180 is driven. The converters 140, 180 may be controllable converters. Alternatively, however, the order may be fixed and thus permanently programmed so that the inputs designated n ₁ and n ₂ are not present in this embodiment.

Fig. 1b shows an apparatus for decompressing compressed sound field data 20 having first harmonic components of a first order and one or more second harmonic components of a second order, such as those of FIG Fig. 1a be issued at 141, 182. However, the decompressed sound field data need not necessarily be the harmonic components 141, 142 in "raw format". Instead, could be in Fig. 1a a lossless entropy coder, such as a Huffman coder or an arithmetic coder, to further reduce the number of bits ultimately needed to represent the harmonic components. Then, the data stream 20 fed to an input interface 200 would consist of entropy-coded harmonic components and, if necessary, page information, as described with reference to FIGS Fig. 3a is pictured. In this case, a respective entropy decoder would be provided at the output of the input interface 200, which is connected to the entropy encoder on the encoder side, ie Fig. 1a is adjusted. Thus, the first harmonic components of the first order 201 and the second harmonic components of the second order 202, as shown in FIG Fig. 1b optionally entropy-coded or else entropy-decoded or indeed the harmonic components in "raw" form, as indicated at 141, 182 in Fig. 1a present,

Both groups of harmonic components are fed to a decoder / combiner 240. The block 240 is configured to decompress the compressed sound field data 201, 202 using a combination of the first portion and the second portion and using a translation from a harmonic component representation into a time domain representation, and finally to provide the decompressed representation of the sound field as shown at 240. The decoder 240, which may be formed, for example, as a signal processor, is thus designed, on the one hand, to implement a conversion into the time domain from the spherical-harmonic component region and, on the other hand, to perform a combination. However, the order between implementation and combination may be different as regards Fig. 1d, Fig. 1e or Fig. 2c, 2d for different examples.

Fig. 1c shows an apparatus for compressing sound field data of a region according to an embodiment, in which the splitter 100 is formed as a time divider 100a. In particular, the scheduler 100a is one implementation of the divider 100 of Fig. 1a is configured to divide the sound field data into a first portion comprising first reflections in the area and a second portion comprising second reflections in the area, the second reflections occurring later in time than the first reflections. Based on Fig. 4 Thus, the first portion 101 output from the block 100a represents the impulse response portion 310 of FIG Fig. 4 while the second late portion represents the impulse response portion 320 of FIG Fig. 4 represents. The timing of the division may be, for example, 100 ms. However, other possibilities of temporal distribution exist, such as earlier or later. Preferably, the division is made where the discrete reflections change to diffuse reflections. This can be a different time, depending on the room, and concepts exist to create the best layout here. On the other hand, the division into an early and a late portion can also be carried out depending on an available data rate, in such a way that the distribution time is made smaller and smaller the less bit rate there is. This is favorable in terms of the bit rate, because then as large a proportion of the impulse response with a low order is converted into the harmonic component range.

The converter, defined by blocks 140 and 180 in FIG Fig. 1c is thus configured to convert the first portion 101 and the second portion 102 into harmonic components, the converter in particular converting the second portion into one or more harmonic components 182 having a second order and the first portion 101 into harmonic components 141 with a first order, the first order being higher than the second order, to finally obtain the compressed sound field which is finally outputable from an output interface 190 for transmission and / or storage purposes.

Fig. 1d shows an implementation of the decompressor for the example of the time distribution. In particular, the decompressor is configured to perform the compressed sound field data using a combination of the first portion 201 with the first reflections and the second portion 202 with the late reflections and a conversion from the harmonic component range to the time domain. Fig. 1d shows an implementation in which the combination takes place after the conversion. Fig. 1e shows an alternative implementation in which the combination takes place before the conversion. Specifically, the converter 241 is configured to convert harmonic components of the high order into the time domain while the converter 242 is configured to form the harmonic components of the low order to implement in the time domain. With regard Fig. 4 Thus, the output of converter 241 provides something corresponding to region 210, while converter 242 provides something corresponding to region 320, but due to the lossy compression, the portions at the output of bridge 241, 242 are not identical to portions 310, 320 are. In particular, however, an at least perceptual similarity or identity of the portion at the output of the block 241 to the portion 310 of FIG Fig. 4 while the portion at the output of the block 242 corresponding to the late portion 320 of the impulse response will have significant differences and thus approximate only the course of the impulse response. However, these deviations are not critical to human directional perception, because human directional perception is hardly or not based on the late contribution or diffuse reflections of the impulse response anyway.

Fig. 1e FIG. 12 shows an alternative implementation in which the decoder first comprises the combiner 245 and then the converter 244. The individual harmonic components are used in the in Fig. 1e Then, the result of the addition is converted to finally obtain a time domain representation. In contrast, when running in Fig. 1d a combination does not consist in an addition, but in a serialization, in that the output of the block 241 will be arranged earlier in a decompressed impulse response than the output of the block 242 to again one Fig. 4 To obtain corresponding impulse response, which can then be used for further purposes, such as an auralization in a preparation of sound signals with the desired spatial impression.

Fig. 2a shows an alternative implementation of the present invention, in which a division in the frequency domain is made. In particular, the splitter 100 is of Fig. 1a in the embodiment of Fig. 2a implemented as a filter bank to filter at least part of the sound field data to obtain sound field data in different filter bank channels 101, 102. The filter bank receives in an embodiment in which the temporal distribution of Fig. 1a is not implemented, both the early and the late portion, while in an alternative embodiment, only the early portion of the sound field data is fed into the filter bank, while the late portion is not spectrally decomposed further.

Subordinate to the analysis filter bank 100b is the converter, which may be formed of sub-converters 140a, 140b, 140c. The converter 140a, 140b, 140c is configured to convert the sound field data in different filter bank channels using different orders for different filter bank channels to obtain one or more harmonic components for each filter bank channel. In particular, the converter is configured to perform a first-order conversion for a first filterbank channel having a first center frequency, and to perform a second-order conversion for a second filterbank channel having a second center frequency, the first order being higher than the second order and wherein the first center frequency, ie, f _n , is higher than the second center frequency f ₁ to finally obtain the compressed sound field representation. Generally, depending on the embodiment, a lower order may be used for the lowest frequency band than for a middle frequency band. However, depending on the implementation, the highest frequency band, as in the case of the Fig. 2a In the embodiment shown, the filter bank channel with the center frequency f _n is not necessarily implemented with a higher order than, for example, a middle channel. Instead, in the areas in which the direction perception is highest, the highest order can be used, as in the other areas, which may include, for example, a certain high frequency range, the order is lower, because in these areas, the direction perception of the human hearing is lower.

Fig. 2b shows a more detailed implementation of the analysis filter bank 100b. This includes at the in Fig. 2b shown embodiment, a bandpass filter and further has downstream decimator 100c for each filter bank channel. For example, if a filter bank consisting of bandpass filters and decimators having 64 channels is used, each decimator can be decimated by a factor of 1/64, so that the total number of digital samples at the output of the decimators over all channels adds up to the number of samples of one Blocks of sound field data in the time domain, which has been decomposed by the filter bank. An exemplary filter bank may be a real or complex QMF filter bank. Each subband signal, preferably the early portions of the impulse responses, is then converted into harmonic components by means of the converters 140a to 140c in analogy to Fig. 2a implemented, finally, for various subband signals of the sound field description to obtain a description with cylindrical or preferably spherical harmonic components having different orders for different subband signals, that is, a different number of harmonic components.

Fig. 2c and Fig. 2d again show different implementations of the decompressor, as in Fig. 1b is shown, so a different order of the combination and subsequent implementation in Fig. 2c or the initial implementation and the subsequent combination, as described in Fig. 2d is shown. In particular, the decompressor 240 includes Fig. 1b at the in Fig. 2c Again, a combiner 245, which performs an addition of the different harmonic components from the various subbands, in order to then obtain an overall representation of the harmonic components, which are then converted with the converter 244 in the time domain. Thus, the inputs to combiner 245 are in the harmonic component spectral region, while the output of combiner 345 is a harmonic component region representation, which is then converted by converter 244 to the time domain.

At the in Fig. 2d In the alternative embodiment shown, the individual harmonic components for each subband are first converted into the spectral range by different converters 241a, 241b, 241c, so that the output signals of the blocks 241a, 241b, 241c correspond to the output signals of the blocks 140a, 140b, 140c of FIG Fig. 2a or Fig. 2b correspond. These subband signals then become in a downstream synthesis filter bank which, in the case of an encoder side downsampling (block 100c of FIG Fig. 2b ) may also have a high-touch function, that is, an upsampling function, processed. The synthesis filterbank then sets the combiner function of the decoder 240 of FIG Fig. 1b At the output of the synthesis filter bank is thus the decompressed sound field representation, which can be used for auralization, as will be shown.

Fig. 1f shows an example of the decomposition of impulse responses into harmonic components of different orders. The late sections are not spectrally decomposed but implemented in total with the zeroth order. The early sections of the impulse responses are spectrally decomposed. For example, the lowest band is processed with the first order, while the next band is already processed with the fifth order and the last band, because it is most important for direction / space perception, with the highest order, in this example with order 14, is processed.

Fig. 3a Figure 4 shows the entire encoder / decoder scheme or the entire compressor / decompressor scheme of the present invention.

In particular, in the Fig. 3a shown embodiment of the compressor not only the functionalities of Fig. 1a , denoted by 1 or PENC, but also a decoder PDEC2, as in Fig. 1b can be trained. In addition, the compressor also comprises a controller CTRL4, which is designed to compare decompressed sound field data obtained by the decoder 2 with original sound field data taking into account a psychoacoustic model, such as the model PEAQ, which has been standardized by the ITU.

Thereafter, the controller 4 generates optimized parameters for the division, such as the time distribution, the frequency distribution in the filter bank or optimized parameters for the orders in the individual converters for the different parts of the sound field data, if these converters are designed to be controllable.

Control parameters, such as split information, filter bank parameters, or orders, may then be transmitted along with a bit stream having the harmonic components to a decompressor denoted 2 in Fig. 3a is shown. The compressor 11 thus consists of the control block CTRL4 for the codec control as well as a parameter encoder PENC1 and the parameter decoder PDEC2. The inputs 10 are data from microphone array measurements. The control block 4 initializes the encoder 1 and provides all the parameters for encoding the array data. In PENC block 1, the data is processed in accordance with the methodology described above, in the time domain and in the frequency domain, and made available for data transmission.

Fig. 3b shows the scheme of data en- and decoding. The input data 10 are first decomposed by the splitter 100a into an early 101 and a late sound field 102. The early sound field 101 is decomposed by means of an n-band filter bank 100b into its spectral components f ₁ ... F _n , each of which has a spherical harmonic order adapted to the human ear (x-order SHD = Spherical Harmonics Decomposition). This spherical harmonic decomposition is a preferred embodiment, but can be performed by any sound field decomposition that generates harmonic components. Since the decomposition into spherical harmonic components in each band depending on Order requires different calculation times, it is preferred to correct the time offsets in a delay line with delay blocks 306, 304. Thus, the frequency range in the reconstruction block 245, which is also referred to as a combiner, is reconstructed and recombined with the late sound field in the further combiner 243 after it has been computed with a low-order listener fit.

The control block CTRL4 of Fig. 3a includes a room acoustic analysis module and a psychoacoustics module. The control block analyzes both the input data 10 and the output data of the decoder 2 of FIG Fig. 3a to encode the encoding parameters, also called page information 300 in FIG Fig. 3a or directly provided in the compressor 11 to the encoder PENC1, adaptively adapt. From the input signals 10, room acoustic parameters are extracted which specify the initial parameters of the coding with the parameters of the array configuration used. These include both the time of separation between early and late sound field, also referred to as "mixing time", and the parameters for the filter bank, such as corresponding orders of spherical harmonics. The output, which may be in the form of binaural impulse responses, for example, as output from combiner 243, is fed into a psychoacoustic module with an auditory model that evaluates the quality and adjusts the encoding parameters accordingly. Alternatively, the concept can also work with static parameters. Then the control module CTRL4 and the PEDC module 2 on the encoder or compressor side 11 is omitted.

The invention is advantageous in that data and computational complexity in the processing and transmission of circular and spherical array data are reduced as a function of human hearing. It is also advantageous that the data thus processed can be integrated into existing compression methods and thus allow additional data reduction. This is advantageous in band-limited transmission systems, such as for mobile terminals. Another advantage is the possible real-time processing of the data in the spherical harmonic range even at high orders. The present invention can be used in many fields, and in particular in the areas where the acoustic sound field is represented by means of cylindrical or spherical harmonics. This is done eg in the sound field analysis by means of circular or ball arrays. If the analyzed sound field is to be auralized, the concept of the present invention can be used. For devices for the simulation of rooms Databases used to store existing spaces. Here, the inventive concept allows a space-saving and high-quality storage. There are reproduction methods based on spherical surface functions, such as Higher Order Ambisonics or binaural synthesis. Here, the present invention provides a reduction of computation time and data overhead. This can be of particular advantage, in particular with regard to data transmission, for example in the case of teleconferencing systems.

Fig. 5 shows an implementation of a converter 140 or 180 with adjustable order or with at least different order, which may also be fixed.

The converter includes a time-frequency transform block 502 and a downstream space transform block 504. The space transform block 504 is configured to operate in accordance with computation rule 508. In the calculation rule, n is the order. The calculation rule 508 is solved only once, if the order is zero, depending on the order, or is solved more often if the order, for example, goes to order 5 or, in the example described above, to order 14. In particular, the time-frequency transform element 502 is configured to transform the impulse responses on the input lines 101, 102 into the frequency domain, preferably employing the fast Fourier transform. Furthermore, then only the half-page spectrum is forwarded to reduce the computational effort. Then, a spatial Fourier transform is performed in the block space transformation 504 as described in US Pat Fourier Acoustics, Sound Radiation and Nearfield Acoustic Holography, Academic Press, 1999 by Earl G. Williams is described. Preferably, the spatial transformation 504 is optimized for sound field analysis while providing high numerical accuracy and fast computation speed.

Fig. 6 Figure 4 shows the preferred implementation of a harmonic component range converter into the time domain where, as an alternative, a planar wave processor and beamforming 602 are illustrated as an alternative to an inverse space transform implementation 604. The outputs of both blocks 602, 604 may alternatively are fed to a block 606 for generating impulse responses. Inverse space transformation 604 is configured to undo the Hin transformation in block 504. Alternatively, the planar wave decomposition and beamforming in block 606 results in a large amount of decomposition directions can be processed evenly, which is favorable for rapid processing, in particular for visualization or auralization. Preferably, the block 602 receives radial filter coefficients and, depending on the implementation, additional beamforming or beamforming coefficients. These can either have a constant directionality or be frequency-dependent. Alternative inputs to block 602 may be modal radial filters, and in particular spherical arrays or different configurations, such as an omnidirectional open-sphere microphone, an open sphere with cardioid microphones, and a rigid sphere with omnidirectional microphones. The impulse response generation block 606 generates impulse responses or time domain signals from either block 602 or block 604. This block recombines in particular the previously omitted negative portions of the spectrum, performs a fast inverse Fourier transform and allows for resampling the original sample rate if the input signal was down-sampled at one location. Furthermore, a window option can be used.

Details of the functionality of the blocks 502, 504, 602, 604, 606 are in the technical publication " SofiA Sound Field Analysis Toolbox "by Bernschütz et al., ICSA - International Conference on Spatial Audio, Detmold, 10 to 13.11.2011 , this reference is incorporated herein by reference in its entirety.

The block 606 may further be configured to output the complete set of decompressed impulse responses, such as the lossy impulse responses, in which case the block 608 would again output, for example, 350 impulse responses. However, depending on the auralization, it is preferred to output only the impulse responses ultimately required for rendering, which may be accomplished by a block 608 that provides selection or interpolation for a particular rendering scenario. For example, if stereo reproduction is desired, as shown in block 616, depending on the placement of the two stereo speakers, the impulse response selected from the 350, for example, recovered impulse responses will be selected, corresponding to the spatial direction of the corresponding stereo speaker. With this impulse response, a prefilter of the corresponding loudspeaker is then set, such that the prefilter has a filter characteristic which corresponds to this impulse response. Then, an audio signal to be reproduced is fed to the two speakers via the corresponding pre-filters and reproduced to finally produce the desired spatial impression for a stereo auralization.

If among the available impulse responses there is not an impulse response in a particular direction in which a loudspeaker is located in the actual rendering scenario, preferably the two or three nearest impulse impulses are used and interpolation is performed.

In an alternative embodiment in which the rendering or auralization takes place by wave-field synthesis 612, it is preferred to perform a replay of early and late reflections via virtual sources, as described in US Pat PH. D.-work "Spatial Sound Design based on Measured Room Impulse Responses" by Frank Melchior at the TU Delft from the year 2011 is shown in detail, this reference is also incorporated herein by reference in its entirety.

In particular, the reflections of a source in wave field synthesis playback 612 are represented by four impulse responses at certain positions for the early reflections and eight impulse responses at certain positions for the late reflections. Selection block 608 then selected the 12 impulse responses for the 12 virtual positions. Subsequently, these impulse responses are fed along with the associated positions in a wavefield synthesis renderer, which may be located at block 612, and the wavefield synthesis renderer then uses these impulse responses to compute the loudspeaker signals for the speakers actually present to make them then map the corresponding virtual sources. This calculates a separate pre-filter for each loudspeaker in the Wave Field Synthesis Playback System, which is then used to filter an audio signal to be ultimately reproduced before it is output from the loudspeaker to achieve adequate high quality spatial effects.

An alternative implementation of the present invention is a generation of a headphone signal, ie in a binaural application, in which the spatial impression of the area is to be generated via the headphone reproduction.

Although primarily impulse responses have been presented as sound field data in the foregoing, any other sound field data, for example sound field data by magnitude and vector, ie with regard to eg sound pressure and sound velocity at certain positions in space, may also be used. Also, these sound field data may become more important and less important in terms of human directional perception divided and converted into harmonic components. The sound field data may also include any type of impulse responses, such as Head-Related Transfer Functions (HRTF) functions or Binaural Room Impulse Responses (BRIR) functions or impulse responses, each from a discrete point to a predetermined position in the area.

Preferably, a space is scanned with a ball array. Then the sound field is a set of impulse responses. In the time domain, the sound field is split into its early and late parts. Subsequently, both parts are decomposed into their spherical or cylindrical harmonic components. Since the relative direction information is present in the early sound field, a higher order of the spherical harmonic is calculated here than in the late sound field, which is sufficient for a low order. The early part is relatively short, for example 100 ms, and is represented accurately, that is, with many harmonic components, while the late part is, for example, 100 ms to 2 s or 10 s long. However, this late part is represented with less or only a single harmonic component.

A further data reduction results from the splitting of the early sound field into individual bands before the representation as spherical harmonics. For this purpose, after the separation in the time domain in early and late sound field, the early sound field is decomposed into its spectral components by means of a filter bank. By subsampling the individual frequency bands, a data reduction is achieved which significantly accelerates the calculation of the harmonic components. In addition, for each frequency band, a sufficiently early order is used as a function of human directional perception. Thus, for low frequency bands in which human directional perception is low, low orders or even zero order for the lowest frequency band are sufficient, while for high bands higher orders are needed to the maximum reasonable order with respect to the accuracy of the measured sound field. On the decoder or decompressor side the complete spectrum is reconstructed. Subsequently, early or late sound field are combined again. The data is now ready for auralization.

Although some aspects have been described in the context of a device, it will be understood that these aspects also constitute a description of the corresponding method such that a block or device of a device is also to be understood as a corresponding method step or feature of a method step is. Similarly, aspects described in connection with or as a method step also represent a description of a corresponding block or detail or feature of a corresponding device. Some or all of the method steps may be performed by a hardware device (or using a hardware device). Apparatus), such as a microprocessor, a programmable computer or an electronic circuit. In some embodiments, some or more of the most important method steps may be performed by such an apparatus.

Depending on particular implementation requirements, embodiments of the invention may be implemented in hardware or in software. The implementation may be performed using a digital storage medium, such as a floppy disk, a DVD, a Blu-ray Disc, a CD, a ROM, a PROM, an EPROM, an EEPROM or FLASH memory, a hard disk, or other magnetic disk or optical memory are stored on the electronically readable control signals that can cooperate with a programmable computer system or cooperate such that the respective method is performed. Therefore, the digital storage medium can be computer readable.

Thus, some embodiments according to the invention include a data carrier having electronically readable control signals capable of interacting with a programmable computer system such that one of the methods described herein is performed.

In general, embodiments of the present invention may be implemented as a computer program product having a program code, wherein the program code is operable to perform one of the methods when the computer program product runs on a computer.

The program code can also be stored, for example, on a machine-readable carrier.

Other embodiments include the computer program for performing any of the methods described herein, wherein the computer program is stored on a machine-readable medium.

In other words, an embodiment of the method according to the invention is thus a computer program which has a program code for performing one of the methods described herein when the computer program runs on a computer.

A further embodiment of the inventive method is thus a data carrier (or a digital storage medium or a computer-readable medium) on which the computer program is recorded for carrying out one of the methods described herein.

A further embodiment of the method according to the invention is thus a data stream or a sequence of signals, which represent the computer program for performing one of the methods described herein. The data stream or the sequence of signals may be configured, for example, to be transferred via a data communication connection, for example via the Internet.

Another embodiment includes a processing device, such as a computer or a programmable logic device, that is configured or adapted to perform one of the methods described herein.

Another embodiment includes a computer on which the computer program is installed to perform one of the methods described herein.

Another embodiment according to the invention comprises a device or system adapted to transmit a computer program for performing at least one of the methods described herein to a receiver. The transmission can be done for example electronically or optically. The receiver may be, for example, a computer, a mobile device, a storage device or a similar device. For example, the device or system may include a file server for transmitting the computer program to the recipient.

In some embodiments, a programmable logic device (eg, a field programmable gate array, an FPGA) may be used to perform some or all of the functionality of the methods described herein. In some embodiments, a field programmable gate array may include a Microprocessor cooperate to perform any of the methods described herein. In general, in some embodiments, the methods are performed by any hardware device. This may be a universal hardware such as a computer processor (CPU) or hardware specific to the process, such as an ASIC.

The embodiments described above are merely illustrative of the principles of the present invention. It will be understood that modifications and variations of the arrangements and details described herein will be apparent to others of ordinary skill in the art. Therefore, it is intended that the invention be limited only by the scope of the appended claims and not by the specific details presented in the description and explanation of the embodiments herein.

Claims (21)

1. Apparatus for compressing sound field data (10) of an area, comprising:

   a divider (100) for dividing the sound field data into a first portion (101) and into a second portion (102); and

   a converter (140, 180) for converting the first portion (101) and the second portion (102) into harmonic components (141, 182) of a sound field description, wherein the converter (140, 180) is configured to convert the second portion (102) into one or several harmonic components (141) of a second order, and to convert the first portion (101) into harmonic components of a first order, wherein the first order is higher than the second order, to obtain the compressed sound field data,

   wherein the divider (100) is configured to perform spectral division and comprises a filterbank (100b) for filtering at least part of the sound field data (10) for obtaining sound field data in different filterbank channels (140a, 140b, 140c), and

   wherein the converter is configured to compute, for a subband signal from a first filterbank channel (140c), which represents the first portion (101), of the different filterbank channels (140a, 140b, 140c), the harmonic components of the first order, and to compute, for a subband signal from a second filterbank channel (140a), which represents the second portion (102), of the different filterbank channels (140a, 140b, 140c), the harmonic components of the second order, wherein a center frequency (f_n) of the first filterbank channel (140a) is higher than a center frequency (f₁) of the second filterbank channel (140c).
2. Apparatus according to claim 1,
   wherein the converter (140, 180) is configured to compute the harmonic components of the first order, which is higher than the second order, for the first portion, which is more important for directional perception of the human hearing than the second portion.
3. Apparatus according to claims 1 or 2,
   wherein the divider (100) is configured to divide the sound field data (10) into the first portion including first reflections in the area and into the second portion including second reflections in the area, wherein the second reflections occur later in time than the first reflections.
4. Apparatus according to one of the previous claims,
   wherein the divider (100) is configured to divide the sound field data into the first portion including first reflections in the area and into the second portion including second reflections in the area, wherein the second reflections occur later in time than the first reflections, and wherein the divider (100) is further configured to decompose the first portion into spectral portions (101, 102) and to convert the spectral portions each into one or several harmonic components of different orders, wherein an order for a spectral portion with a higher frequency band is higher than an order for a spectral portion in a lower frequency band.
5. Apparatus according to one of the previous claims, further comprising an output interface (190) for providing the one or several harmonic components (182) of the second order and the harmonic components of the first order (141) together with side information (300) comprising an indication on the first order or the second order for transmission and storage.
6. Apparatus according to one of the previous claims,
   wherein the sound field data describe a three-dimensional area and the converter is configured to compute cylindrical harmonic components as the harmonic components, or
   wherein the sound field data (10) describe a three-dimensional area and the converter (140, 180) is configured to compute spherical harmonic components as the harmonic components.
7. Apparatus according to one of the previous claims,
   wherein the sound field data exist as a first number of discrete signals,
   wherein the converter (140, 180) for the first portion (101) and the second portion (102) provides a second total number of harmonic components, and
   wherein the second total number of harmonic components is smaller than the first number of discrete signals.
8. Apparatus according to one of the previous claims,
   wherein the divider (100) is configured to use, as sound field data (10), a plurality of different impulse responses that are allocated to different positions in the area.
9. Apparatus according to claim 8,
   wherein the impulse responses are head-related transfer functions (HRTF) or binaural room impulse responses (BRIR) functions or impulse responses of a respective discrete point in the area to a predetermined position in the area.
10. Apparatus according to one of the previous claims, further comprising:

    a decoder (2) for decompressing the compressed sound field data by using a combination of the first and second portions and by using a conversion from a harmonic component representation into a time domain representation for obtaining a decompressed representation; and

    a control (4) for controlling the divider (100) or the converter (140, 180) with respect to the first or second order, wherein the control (4) is configured to compare, by using a psychoacoustic module, the decompressed sound field data with the sound field data (10) and to control the divider (100) or the converter (140, 180) by using the comparison.
11. Apparatus according to claim 10,
    wherein the decoder is configured to convert the harmonic components of the second order and the harmonic components of the first order (241, 242) and to then perform a combination of the converted harmonic components, or
    wherein the decoder (2) is configured to combine the harmonic components of the second order and the harmonic components of the first order (245) and to convert a result of the combination in the combiner (245) from a harmonic component domain into the time domain (244).
12. Apparatus according to claim 10,
    wherein the decoder is configured to convert harmonic components of different spectral portions with different orders (140a, 140b),
    to compensate different processing times for different spectral portions (304, 306), and
    to combine spectral portions of the first portion converted into a time domain with the spectral components of the second portion converted into the time domain by serially arranging the same.
13. Apparatus for decompressing compressed sound field data comprising first harmonic components (141) of a sound field description up to a first order and one or several second harmonic components (182) of a sound filed description up to a second order, wherein the first order is higher than the second order, comprising:

    an input interface (200) for obtaining the compressed sound field data; and

    a processor (240) for processing the first harmonic components (201) and the second harmonic components (202) by using a combination of the first and the second portion and by using a conversion of a harmonic component representation into a time domain representation to obtain a decompressed illustration, wherein the first portion is represented by the first harmonic components and the second portion by the second harmonic components,

    wherein the first harmonic components (HK_n) of the first order represent a first spectral domain, and the one or the several harmonic components (HK₁) of the second order represent a different spectral domain,

    wherein the processor (240) is configured to convert the harmonic components (HK_n) of the first order into the spectral domain (241a) and to convert the one or the several second harmonic components (HK₁) of the second order into the spectral domain (241c), and to combine the converted harmonic components by means of a synthesis filterbank (245) to obtain a representation of sound field data in the time domain.
14. Apparatus according to claim 13, wherein the processor (240) comprises:

    a combiner (245) for combining the first harmonic components and the second harmonic components to obtain combined harmonic components; and

    a converter (244) for converting the combined harmonic components into the time domain.
15. Apparatus according to claim 13, wherein the processor comprises:

    a converter (241, 242) for converting the first harmonic components and the second harmonic components into the time domain; and

    a combiner (243, 245) for combining the harmonic components converted into the time domain for obtaining the decompressed sound field data.
16. Apparatus according to one of claims 13 to 15,
    wherein the processor (240) is configured to obtain information on a reproduction arrangement (610, 612, 614), and
    wherein the processor (240) is configured to compute the decompressed sound field data (602, 604, 606) and to select, based on the information on the reproduction arrangement, part of the sound field data of the decompressed sound field data for reproduction purposes (608), or
    wherein the processor is configured to compute only a part of the decompressed sound field data necessitated for the reproduction arrangement.
17. Apparatus according to one of claims 13 to 16,
    wherein the first harmonic components of the first order represent early reflections of the area and the second harmonic components of the second order represent late reflections of the area, and
    wherein the processor (240) is configured to add the first harmonic components and the second harmonic components and to convert a result of the addition into the time domain for obtaining the decompressed sound field data.
18. Apparatus according to one of claims 13 to 17,
    wherein the processor is configured to perform, for the conversion, an inverse room transformation (604) and an inverse Fourier transformation (606).
19. Method for compressing sound field data (10) of an area, comprising the steps of:

    dividing (100) the sound field data into a first portion (101) and into a second portion (102), and

    converting (140, 180) the first portion (101) and the second portion (102) into harmonic components (141, 182) of a sound field description, wherein the second portion (102) is converted into one or several harmonic components (141) of a second order, and wherein the first portion (101) is converted into harmonic components of a first order, wherein the first order is higher than the second order, to obtain the compressed sound field data,

    wherein dividing (100) comprises spectral division by filtering with a filterbank (100b) for filtering at least part of the sound field data (10) for obtaining sound field data in different filterbank channels (140a, 140b, 140c), and

    wherein converting represents a computation of the harmonic components of the first order for a subband signal from a first filterbank channel (140c), which represents the first portion (101), of the different filterbank channels (140a, 140b, 140c), and a computation of the harmonic components of the second order for a subband signal from a second filterbank channel (140a), which represents the second portion (102), of the different filterbank channels (140a, 140b, 140c), wherein a center frequency (f_n) of the first filterbank channel (140a) is higher than a center frequency (f₁) of the second filterbank channel (140c).
20. Method for decompressing compressed sound field data comprising first harmonic components (141) of a sound field description up to a first order and one or several second harmonic components (182) of a sound field description up to a second order, wherein the first order is higher than the second order, comprising steps of:

    obtaining (200) the compressed sound field data; and

    processing (240) the first harmonic components (201) and the second harmonic components (202) by using a combination of the first and second portions and by using a conversion from a harmonic component representation into a time domain representation to obtain a decompressed representation, wherein the first portion is represented by the first harmonic components and the second portion by the second harmonic components,

    wherein the first harmonic components (HK_n) of the first order represent a first spectral domain, and the one or the several harmonic components (HK₁) of the second order represent a different spectral domain,

    wherein processing (240) comprises converting the first harmonic components (HK_n) of the first order into the spectral domain and converting the one or the several second harmonic components (HK₁) of the second order into the spectral domain and combining the converted harmonic components by means of a synthesis filterbank (245) to obtain a representation of sound field data in the time domain.
21. Computer program for performing a method according to one of claims 19 to 20 when the method runs on a computer.

Images