Analysis Followed by Partial Synthesis

Embodiments relate to an audio processor/method for processing an audio signal to obtain a Bubband representation of the audio signal. Further embodiments relate to an audio processor/method for processing a subband representation of an audio signal to obtain the audio signal. Some embodiments relate to time domain aliasing reduction In subbands of non-uniform orthogonal fllterbanks based on MDCT (MDCT = modified discrete cosine transform) analysis/synthesis, e.g., in subbands of non-uniform orthogonal MDCT fllterbanks.

MDCT Is widely used In audio coding applications due to lis properties like good energy compaction and orthogonality when used In a lapped fashion. However, MDCT exhibits a uniform time-frequency resolution [J. Princen, A. Johnson, and A. Bradley, "Subband/transform coding using filter bank designs based on time domain aliasing cancellation," In Acoustics, Speech, and Signal Processing, IEEE International Conference on ICASSP '87., Apr 1987, vol. 12, pp. 2181-2164]. When doing perceptually motivated audio processing, however, a non-uniform time-frequency resolution may be a more desirable representation.

One way of designing a non-uniform transform Is the repeated application of one of several uniform transforms.

For subband merging first a long transform is applied, transforming the signal from the temporal to the spectral domain. The result Is a spectrum with high spectral but tow temporal resolution. Afterwards several spectral bins are transformed back to the temporal domain. This Increases the temporal resolution while sacrificing spectral resolution in that selected subband.

Subband splitting Is the complementary operation: First a short transform Is applied. The result is a spectrum with tow spectral but high temporal resolution. Afterwards, the spectral bins of two or more adjacent transform frames are transformed again, Increasing their spectral resolution at the cost of temporal resolution.

These steps can be mixed and repeated at will. The choice of transform can be arbitrary, however the same or a similar transforms for each step is usually chosen.

There exist numerous ways of facilitating non-uniform time-frequency transforms:

Using two consecutive fast Fourier transforms, there exists the ERBLet transform, a subband mergin transform with an ERB frequency scale ΓΤ. Necciari, P. Balazs, N. Holighaus, and P.L. Sondergaard, "The erblet transform: An auditory-based time-frequency representation with perfect reconstruction," in Acoustics, Speech and Signal Processing (ICASSP), 2013 IEEE International Conference on, May 2013, pp. 498-502]. Recently, the same authors expanded their approach to a discrete cosine transform type 4 (DCT4) spectrum and a MDCT subband merging transform [Olivier Oerrien, Thibaud Necciari, and Peter Balazs, "A quasi-orthogonal, Invertible, and perceptually relevant time-frequency transform for audio coding," in EUSIPCO, Nice, France, Aug. 2015].

However, both approaches were designed to require very long, overlapping transform windows with non-critical sampling or even transforming the entire signal in one step. These long transform windows and non-critical sampling prohibit precise time-localization in the transform domain and make them unsuitable for coding applications due to a large look ahead and high redundancy.

A subband merging technique using MDCT and butterfly elements to combine selected coefficients of one MDCT frame were introduced in [J. Mau, J. Valot, and D. Minaud, "Time-varying orthogonal filter banks without transient filters," in Proceedings of the Acoustics, Speech, and Signal Processing, 1995. On International Conference - Volume 02, Washington, DC, USA, 1995, ICASSP '95, pp. 1328-1331, IEEE Computer Society] and generalized to Hadamard matrices in [O.A. Niamut and R. Heusdens, "Flexible frequency decompositions for cosine-modulated filter banks," in Acoustics, Speech, and Signal Processing, 2003. Proceedings. (ICASSP Ό3). 2003 IEEE International Conference on, April 2003, vol. 5, pp. V-449-52 vol.5]. The complementary subband splitting operation was introduced in [Jean-Marc Valin, Gregory Maxwell, Timothy B. Terriberry, and Koen Vos, "High-quality, low-delay music coding in the opus codec," in Audio Engineering Society Convention 135, Oct 2013].

While allowing direct integration into common lapped MDCT transform pipelines, these Butterfly- and Hadamard-based implementations only allow for very limited frequency scale designs with for example sizes constrained to k = 2n with n∈ 0
Additionally, the Hadamard matrix only very roughly approximates the DCT and thus allows for only very limited tempo-spectral-resolution, as will be described in more detail below.

Additionally, while some of these methods use MDCT they do not try to reduce the resulting aliasing in the subbands, producing a smeared temporal compactness of the resulting interbank impulse.

Therefore, it is the object of the present invention to provide a concept that that provides at least one out of an improved temporal compactness of the impulse response, processing arbitrary frequency scales, and reduced redundancy and delay.

This object is solved by the independent claims.

Embodiments provide an audio processor for processing an audio signal to obtain a subband representation of the audio signal. The audio processor comprises a cascaded lapped critically sampled transform stage and a time domain aliasing reduction stage. The cascaded lapped critically sampled transform stage is configured to perform a cascaded lapped critically sampled transform on at least two partially overlapping blocks of samples of the audio signal, to obtain a set of subband samples on the basis of a first block of samples of the audio signal, and to obtain a corresponding set of subband samples on the basis of a second block of samples of the audio signal. The time domain aliasing reduction stage is configured to perform a weighted combination of two corresponding sets of subband samples, one obtained on the basis of the first block of samples of the audio signal and one obtained on the basis on the second block of samples of the audio signal, to obtain an aliasing reduced subband representation of the audio signal.

Further embodiments provide an audio processor for processing a subband representation of an audio signal to obtain the audio signal. The audio processor comprises an inverse time domain aliasing reduction stage and a cascaded inverse lapped critically sampled transform stage. The inverse time domain aliasing reduction stage is configured to perform a weighted (and shifted) combination of two corresponding aliasing reduced subband representations (of different blocks of partially overlapping samples) of the audio signal, to obtain an aliased subband representation, wherein the aliased subband representation is a set of subband samples. The cascaded inverse lapped critically sampled transform stage is configured to perform a cascaded inverse lapped critically sampled transform on the set of subband samples, to obtain a set of samples associated with a block of samples of the audio signal.

According to the concept of the present invention, an additional post-processing stage is added to the lapped critically sampled transform (e.g., MDCT) pipeline, the additional postprocessing stage comprising another lapped critically sampled transform (e.g., MDCT) along the frequency axis and a time domain aliasing reduction along each subband time axis. This allows extracting arbitrary frequency scales from the lapped critically sampled transform (e.g., MDCT) spectrogram with an improved temporal compactness of the impulse response, while introducing no additional redundancy and a reduced lapped critically sampled transform frame delay.

Further embodiments provide a method for processing an audio signal to obtain a subband representation of the audio signal. The method comprises

- performing a cascaded lapped critically sampled transform on at least two partially overlapping blocks of samples of the audio signal, to obtain a set of subband samples on the basis of a first block of samples of the audio signal, and to obtain a corresponding set of subband samples on the basis of a second block of samples of the audio signal; and

- performing a weighted combination of two corresponding sets of subband samples, one obtained on the basis of the first block of samples of the audio signal and one obtained on the basis on the second block of samples of the audio signal, to obtain an aliasing reduced subband representation of the audio signal.

Further embodiments provide a method for processing a subband representation of an audio signal to obtain the audio signal. The method comprises:

- performing a weighted (and shifted) combination of two corresponding aliasing reduced subband representations (of different blocks of partially overlapping samples) of the audio signal, to obtain an aliased subband representation, wherein the aliased subband representation is a set of subband samples; and

- performing a cascaded inverse lapped critically sampled transform on the set of subband samples, to obtain a set of samples associated with a block of samples of the audio signal.

Advantageous implementations are addressed in the dependent claims.

Subsequently, advantageous implementations of the audio processor for processing an audio signal to obtain a subband representation of the audio signal are described.

In embodiments, the cascaded lapped critically sampled transform stage can be a cascaded MDCT (MDCT = modified discrete cosine transform), MDST (MDST = modified discrete sine transform) or MLT (MLT = modulated lapped transform) stage.

In embodiments, the cascaded lapped critically sampled transform stage can comprise a first lapped critically sampled transform stage configured to perform lapped critically sampled transforms on a first block of samples and a second block of samples of the at least two partially overlapping blocks of samples of the audio signal, to obtain a first set of bins for the first block of samples and a second set of bins (lapped critically sampled coefficients) for the second block of samples.

The first lapped critically sampled transform stage can be a first MDCT, MDST or MLT stage.

The cascaded lapped critically sampled transform stage can further comprise a second lapped critically sampled transform stage configured to perform a lapped critically sampled transform on a segment (proper subset) of the first set of bins and to perform a lapped critically sampled transform on a segment (proper subset) of the second set of bins, each segment being associated with a subband of the audio signal, to obtain a set of subband samples for the first set of bins and a set of subband samples for the second set of bins.

The second lapped critically sampled transform stage can be a second MDCT, MDST or MLT stage.

Thereby, the first and second lapped critically sampled transform stages can be of the same type, i.e. one out of MDCT, MDST or MLT stages.

In embodiments, the second lapped critically sampled transform stage can be configured to perform lapped critically sampled transforms on at least two partially overlapping segments (proper subsets) of the first set of bins and to perform lapped critically sampled transforms on at least two partially overlapping segments (proper subsets) of the second set of bins, each segment being associated with a subband of the audio signal, to obtain at least two sets of subband samples for the first set of bins and at least two sets of subband samples for the second set of bins.

Thereby, the first set of subband samples can be a result of a first lapped critically sampled transform on the basis of the first segment of the first set of bins, wherein a second set of subband samples can be a result of a second lapped critically sampled transform on the

basis of the second segment of the first set of bins, wherein a third set of subband samples can be a result of a third lapped critically sampled transform on the basis of the first segment of the second set of bins, wherein a fourth set of subband samples can be a result of a fourth lapped critically sampled transform on the basis of the second segment of the second set of bins. The time domain aliasing reduction stage can be configured to perform a weighted combination of the first set of subband samples and the third set of subband samples, to obtain a first aliasing reduced subband representation of the audio signal, and to perform a weighted combination of the second set of subband samples and the fourth set of subband samples, to obtain a second aliasing reduced subband representation of the audio signal.

In embodiments, the cascaded lapped critically sampled transform stage can be configured to segment a set of bins obtained on the basis of the first block of samples using at least two window functions and to obtain at least two sets of subband samples based on the segmented set of bins corresponding to the first block of samples, wherein the cascaded lapped critically sampled transform stage can be configured to segment a set of bins obtained on the basis of the second block of samples using the at least two window functions and to obtain at least two sets of subband samples based on the segmented set of bins corresponding to the second block of samples, wherein the at least two window functions comprise different window width.

In embodiments, the cascaded lapped critically sampled transform stage can be configured to segment a set of bins obtained on the basis of the first block of samples using at least two window functions and to obtain at least two sets of subband samples based on the segmented set of bins corresponding to the first block of samples, wherein the cascaded lapped critically sampled transform stage can be configured to segment a set of bins obtained on the basis of the second block of samples using the at least two window functions and to obtain at least two sets of subband samples based on the segmented set of bins corresponding to the second block of samples, wherein filter slopes of the window functions corresponding to adjacent sets of subband samples are symmetric.

In embodiments, the cascaded lapped critically sampled transform stage can be configured to segment the samples of the audio signal into the first block of samples and the second block of samples using a first window function, wherein the lapped critically sampled transform stage can be configured to segment a set of bins obtained on the basis of the first block of samples and a set of bins obtained on the basis of the second block of samples using a second window function, to obtain the corresponding subband samples, wherein the first window function and the second window function comprise different window width.

In embodiments, the cascaded lapped critically sampled transform stage can be configured to segment the samples of the audio signal into the first block of samples and the second block of samples using a first window function, wherein the lapped critically sampled transform stage can be configured to segment a set of bins obtained on the basis of the first block of samples and a set of bins obtained on the basis of the second block of samples using a second window function, to obtain the corresponding subband samples, wherein a window width of the first window function and a window width of the second window function are different from each other, wherein the window width of the first window function and the window width of the second window function differ from each other by a factor different from a power of two.

Subsequently, advantageous implementations of the audio processor for processing a subband representation of an audio signal to obtain the audio signal are described.

In embodiments, the inverse cascaded lapped critically sampled transform stage can be an inverse cascaded MDCT (MDCT = modified discrete cosine transform), MDST (MDST = modified discrete sine transform) or MLT (MLT = modulated lapped transform) stage.

In embodiments, the cascaded inverse lapped critically sampled transform stage can comprise a first inverse lapped critically sampled transform stage configured to perform an inverse lapped critically sampled transform on the set of subband samples, to obtain a set of bins associated with a given subband of the audio signal.

The first inverse lapped critically sampled transform stage can be a first inverse MDCT, MDST or MLT stage.

In embodiments, the cascaded inverse lapped critically sampled transform stage can comprise a first overlap and add stage configured to perform a concatenation of a set of bins associated with a plurality of subbands of the audio signal, which comprises a weighted combination of the set of bins associated with the given subband of the audio signal with a set of bins associated with another subband of the audio signal, to obtain a set of bins associated with a block of samples of the audio signal.

In embodiments, the cascaded inverse lapped critically sampled transform stage can comprise a second inverse lapped critically sampled transform stage configured to perform an inverse lapped critically sampled transform on the set of bins associated with the block of samples of the audio signal, to obtain a set of samples associated with the block of samples of the audio signal.

The second inverse lapped critically sampled transform stage can be a second inverse MDCT, MDST or MLT stage.

Thereby, the first and second inverse lapped critically sampled transform stages can be of the same type, i.e. one out of inverse MDCT, MDST or MLT stages.

In embodiments, the cascaded inverse lapped critically sampled transform stage can comprise a second overlap and add stage configured to overlap and add the set of samples associated with the block of samples of the audio signal and another set of samples associated with another block of samples of the audio signal, the block of samples and the another block of samples of the audio signal partially overlapping, to obtain the audio signal.

Embodiments of the present invention are described herein making reference to the appended drawings.

Fig. 1 shows a schematic block diagram of an audio processor configured to process an audio signal to obtain a subband representation of the audio signal, according to an embodiment;

Fig. 2 shows a schematic block diagram of an audio processor configured to process an audio signal to obtain a subband representation of the audio signal, according to a further embodiment;

Fig. 3 shows a schematic block diagram of an audio processor configured to process an audio signal to obtain a subband representation of the audio signal, according to a further embodiment;

Fig. 4 shows a schematic block diagram of an audio processor for processing a subband representation of an audio signal to obtain the audio signal, according to an embodiment;

Fig. 5 shows a schematic block diagram of an audio processor for processing a subband representation of an audio signal to obtain the audio signal, according to a further embodiment;

Fig.6 shows a schematic block diagram of an audio processor for processing a subband representation of an audio signal to obtain the audio signal, according to a further embodiment;

Fig.7 shows in diagrams an example of subband samples (top graph) and the spread of their samples over time and frequency (below graph);

Fig.8 shows in a diagram the spectral and temporal uncertainty obtained by several different transforms;

Fig.9 shows in diagrams shows a comparison of two exemplary impulse responses generated by subband merging with and without TDAR, simple MDCT shortblocks and Hadamard matrix subband merging;

Fig.10 shows a flowchart of a method for processing an audio signal to obtain a subband representation of the audio signal, according to an embodiment;

Fig.11 shows a flowchart of a method for processing a subband representation of an audio signal to obtain the audio signal, according to an embodiment;

Fig.12 shows a schematic block diagram of an audio encoder, according to an embodiment;

Fig.13 shows a schematic block diagram of an audio decoder, according to an embodiment; and

Fig. 14 shows a schematic block diagram of an audio analyzer, according to an embodiment.

Equal or equivalent elements or elements with equal or equivalent functionality are denoted in the following description by equal or equivalent reference numerals.

In the following description, a plurality of details are set forth to provide a more thorough explanation of embodiments of the present invention. However, it will be apparent to one skilled in the art that embodiments of the present invention may be practiced without these specific details. In other instances, well-known structures and devices are shown in block diagram form rather than in detail in order to avoid obscuring embodiments of the present invention. In addition, features of the different embodiments described hereinafter may be combined with each other, unless specifically noted otherwise.

Fig. 1 shows a schematic block diagram of an audio processor 100 configured to process an audio signal 102 to obtain a subband representation of the audio signal, according to an embodiment. The audio processor 100 comprises a cascaded lapped critically sampled transform (LOST) stage 104 and a time domain aliasing reduction (TDAR) stage 106.

The cascaded lapped critically sampled transform stage 104 is configured to perform a cascaded lapped critically sampled transform on at least two partially overlapping blocks 108_1 and 108J2 of samples of the audio signal 102, to obtain a set 110_1,1 of subband samples on the basis of a first block 108_1 of samples (of the at least two overlapping blocks 108_1 and 108_2 of samples) of the audio signal 102, and to obtain a corresponding set 110_2,1 of subband samples on the basis of a second block 108_2 of samples (of the at least two overlapping blocks 108_1 and 108_2 of samples) of the audio signal 102.

The time domain aliasing reduction stage 104 is configured to perform a weighted combination of two corresponding sets 110_1 , 1 and 110_2,1 of subband samples (i.e., subband samples corresponding to the same subband), one obtained on the basis of the first block 108_1 of samples of the audio signal 102 and one obtained on the basis of the second block 108_2 of samples of the audio signal, to obtain an aliasing reduced subband representation 112_1 of the audio signal 102.

In embodiments, the cascaded lapped critically sampled transform stage 104 can comprise at least two cascaded lapped critically sampled transform stages, or in other words, two lapped critically sampled transform stages connected in a cascaded manner.

The cascaded lapped critically sampled transform stage can be a cascaded MDCT (MDCT = modified discrete cosine transform) stage. The cascaded MDCT stage can comprise at least two MDCT stages.

Naturally, the cascaded lapped critically sampled transform stage also can be a cascaded MDST (MDST = modified discrete sine transform) or MLT (MLT = modulated lap transform) stage, comprising at least two MDST or MLT stages, respectively.

The two corresponding sets of subband samples 110.1,1 and 110_2,1 can be subband samples corresponding to the same subband (i.e. frequency band).

Fig. 2 shows a schematic block diagram of an audio processor 100 configured to process an audio signal 102 to obtain a subband representation of the audio signal, according to a further embodiment.

As shown in Fig. 2, the cascaded lapped critically sampled transform stage 104 can comprise a first lapped critically sampled transform stage 120 configured to perform lapped critically sampled transforms on a first block 108_1 of (2M) samples (xI-1(n), 0≤n≤2M-1) and a second block 108_2 of (2M) samples (xi(n), 0≤n≤2M-1) of the at least two partially overlapping blocks 108_1 and 108_2 of samples of the audio signal 102, to obtain a first set 124_1 of (M) bins (LCST coefficients) (XM(k), 0≤k≤M-1) for the first block 108_1 of samples and a second set 124_2 of (M) bins (LCST coefficients) (Xi(k), 0≤k≤M-1) for the second block 108_2 of samples.

The cascaded lapped critically sampled transform stage 104 can comprise a second lapped critically sampled transform stage 126 configured to perform a lapped critically sampled transform on a segment 128_1 ,1 (proper subset) (XV,I-1(k)) of the first set 124_1 of bins and to perform a lapped critically sampled transform on a segment 128_2,1 (proper subset) (Xv,i(k)) of the second set 124_2 of bins, each segment being associated with a subband of the audio signal 102, to obtain a set 110_1,1 of subband samples [yV,I-1(m)] for the first set 124_1 of bins and a set 110_2,1 of subband samples (yv,i(m)) for the second set 124J2 of bins.

Fig. 3 shows a schematic block diagram of an audio processor 100 configured to process an audio signal 102 to obtain a subband representation of the audio signal, according to a further embodiment, in other words, Fig. 3 shows a diagram of the analysis filterbank. Thereby, appropriate window functions are assumed. Observe that for simplicity reasons in Fig. 3 (only) the processing of a first half of a subband frame (y[m], 0 <= m < N/2) (i.e. only the first line of equation (6)) is indicated.

As shown in Fig. 3, the first lapped critically sampled transform stage 120 can be configured to perform a first lapped critically sampled transform 122_1 (e.g., MDCT i-1) on the first block 108_1 of (2M) samples (xI-1(n), 0≤η≤2Μ-1), to obtain the first set 124 J of (M) bins (LCST coefficients) (XI-1(k), 0.sk≤M-1) for the first block 108_1 of samples, and to perform a second lapped critically sampled transform 122_2 (e.g., MDCT i) on the second block 108_2 of (2M)

samples (xI(η), 0≤η≤2Μ-1), to obtain a second set 124.2 of (M) bins (LOST coefficients) (XJ(k), 0≤k≤M-1) for the second block 108.2 of samples.

In detail, the second lapped critically sampled transform stage 126 can be configured to perform lapped critically sampled transforms on at least two partially overlapping segments 128.1,1 and 128.1,2 (proper subsets) (Xv.,I-1(k)) of the first set 124.1 of bins and to perform lapped critically sampled transforms on at least two partially overlapping segments 128.2,1 and 128.2,2 (proper subsets) (Xv.,I-1(k)) of the second set of bins, each segment being associated with a subband of the audio signal, to obtain at least two sets 110.1,1 and 110.1,2 of subband samples (9v,I-1(m)) for the first set 124.1 of bins and at least two sets

110.2.1 and 110.2,2 of subband samples
) for the second set 124.2 of bins.

For example, the first set 110.1,1 of subband samples can be a result of a first lapped critically -sampled transform 132.1,1 on the basis of the first segment 132.1,1 of the first set 124.1 of bins, wherein the second set 110.1,2 of subband samples can be a result of a second lapped critically sampled 132.1,2 transform on the basis of the second segment

128.1.2 of the first set 124.1 of bins, wherein the third set 110.2,1 of subband samples can be a result of a third lapped critically sampled transform 132_2,1 on the basis of the first segment 128.2,1 of the second set 124.2 of obis, wherein the fourth set 110.2,2 of subband samples can be a result of a fourth lapped critically sampled transform 132_2,2 on the basis of the second segment 128.2,2 of the second set 124.2 of bins.

Thereby, the time domain aliasing reduction stage 108 can be configured to perform a weighted combination of the first set 110.1 ,1 of subband samples and the third set 110 _2,1 of subband samples, to obtain a first aliasing reduced subband representation 112.1 (yI-1[m1]) of the audio signal, wherein the domain aliasing reduction stage 108 can be configured to perform a weighted combination of the second set 110.1,2 of subband samples and the fourth est 110J2.2 of subband samples, to obtain a second aliasing reduced subband representation 112 _2 (y2,1[m2]) of the audio signal.

Fig. 4 shows a schematic block diagram of an audio processor 200 for processing a subband representation of an audio signal to obtain the audio signal 102, according to an embodiment. The audio processor 200 comprises an Inverse time domain aliasing reduction (TDAR) stage 202 and a cascaded Inverse lapped critically sampled transform (LOST) stage 204.

The inverse time domain aliasing reduction stage 202 is configured to perform a weighted (and shifted) combination of two corresponding aliasing reduced subband representations 112_1 and 112_2 (yv,i(m), yv,I-1(m)) of the audio signal 102, to obtain an aliased subband representation 110_1 (9v,i(m)), wherein the aliased subband representation is a set 110_1 of subband samples.

The cascaded inverse lapped critically sampled transform stage 204 Is configured to perform a cascaded inverse lapped critically sampled transform on the set 110_1 of subband samples, to obtain a set of samples associated with a block 108_1 of samples of the audio signal 102.

Fig. 5 shows a schematic block diagram of an audio processor 200 for processing a subband representation of an audio signal to obtain the audio signal 102, according to a further embodiment. The cascaded inverse lapped critically sampled transform stage 204 can comprise a first inverse lapped critically sampled transform (LCST) stage 208 and a first overlap and add stage 210.

The first inverse lapped critically sampled transform stage 208 can be configured to perform an inverse lapped critically sampled transform on the set 110_1 ,1 of subband samples, to obtain a set 128_1 ,1 of bins associated with a given subband of the audio signal (£v,i(k)).

The first overlap and add stage 210 can be configured to perform a concatenation of sets of bins associated with a plurality of subbands of the audio signal, which comprises a weighted combination of the set 128_1,1 of bins
associated with the given subband (v) of the audio signal 102 with a set 128_1,2 of bins
associated with another subband (v-1) of the audio signal 102, to obtain a set 124_1 of bins associated with a block 108_1 of samples of the audio signal 102.

As shown in Fig. 5, the cascaded inverse lapped critically sampled transform stage 204 can comprise a second inverse lapped critically sampled transform (LCST) stage 212 configured to perform an inverse lapped critically sampled transform on the set 124_1 of bins associated with the block 108_1 of samples of the audio signal 102, to obtain a set 206_1 , 1 of samples associated with the block 108_1 of samples of the audio signal 102.

Further, the cascaded inverse lapped critically sampled transform stage 204 can comprise a second overlap and add stage 214 configured to overlap and add the set 206_1,1 of samples associated with the block 108_1 of samples of the audio signal 102 and another set 206_2,1 of samples associated with another block 108_2 of samples of the audio signal, the block 108_1 of samples and the another block 108.2 of samples of the audio signal 102 partially overlapping, to obtain the audio signal 102.

Fig. 6 shows a schematic block diagram of an audio processor 200 for processing a subband representation of an audio signal to obtain the audio signal 102, according to a further embodiment. In other words, Fig. 6 shows a diagram of the synthesis filter bank. Thereby, appropriate windows functions are assumed. Observe that for simplicity reasons in Fig. 6 (only) the processing of a first half of a subband frame (y[m], 0 <= m < N/2) (i.e. only the first line of equation (6)) is indicated.

As described above, the audio processor 200 comprises an inverse time domain aliasing reduction stage 202 and an inverse cascades lapped critically sampled stage 204 comprising a first inverse lapped critically sampled stage 208 and a second inverse lapped critically sampled stage 212.

The inverse time domain reduction stage 104 is configured to perform a first weighted and shifted combination 220_1 of a first and second aliasing reduced subband representations y1,I-1[m1] and yI-1[m1] to obtain a first aliased subband representation 110_1,1
wherein the aliased subband representation is a set of subband samples, and to perform a second weighted and shifted combination 220.2 of a third and fourth aliasing reduced subband representations y2,I-1[m1] and y2.i[m1] to obtain a second aliased subband representation 110_2,1 wherein the aliased subband representation is a set of subband samples.

Claims

1. An audio processor (100) for processing an audio signal (102) to obtain a subband representation of the audio signal (102), the audio processor (100) comprising:

a cascaded lapped critically sampled transform stage (104) configured to perform a cascaded lapped critically sampled transform on at least two partially overlapping blocks (108_1;108_2) of samples of the audio signal (102), to obtain a set (110_1,1) of subband samples on the basis of a first block (108_1) of samples of the audio signal (102), and to obtain a corresponding set (110_2,1) of subband samples on the basis of a second block (108_2) of samples of the audio signal (102); and

a time domain aliasing reduction stage (106) configured to perform a weighted combination of two corresponding sets (110_1,1;110_1,2) of subband samples, one obtained on the basis of the first block (108_1) of samples of the audio signal (102) and one obtained on the basis on the second block (108_2) of samples of the audio signal, to obtain an aliasing reduced subband representation (112_1) of the audio signal (102).

2. The audio processor (100) according to claim 1, wherein the cascaded lapped critically sampled transform stage (104) comprises:

a first lapped critically sampled transform stage (120) configured to perform lapped critically sampled transforms on a first block (108_1) of samples and a second block (108_2) of samples of the at least two partially overlapping blocks (108_1;108_2) of samples of the audio signal (102), to obtain a first set (124_1) of bins for the first block (108_1) of samples and a second set (124_2) of bins for the second block (108_2) of samples.

3. The audio processor (100) according to claim 2, wherein the cascaded lapped critically sampled transform stage (104) further comprises:

a second lapped critically sampled transform stage (126) configured to perform a lapped critically sampled transform on a segment (128_1,1) of the first set (124_1) of bins and to perform a lapped critically sampled transform on a segment (128_2,1) of the second set (124_2) of bins, each segment being associated with a subband of the

audio signal (102), to obtain a set (110_1,1) of subband samples for the first set of bins and a set (110_2,1) of subband samples for the second set of bins.

The audio processor (100) according to claim 3, wherein a first set (110_1 ,1) of 4.

subband samples is a result of a first lapped critically sampled transform (132_1,1) on the basis of the first segment (128_1,1) of the first set (124_1) of bins, wherein a second set (110_1,2) of subband samples is a result of a second lapped critically sampled transform (132_1 ,2) on the basis of the second segment (128_1 ,2) of the first set (124_1) of bins, wherein a third set (110_2,1) of subband samples is a result of a third lapped critically sampled transform (132_2,1) on the basis of the first segment (128_2,1) of the second set (128_2,1) of bins, wherein a fourth set (110_2,2) of subband samples is a result of a fourth lapped critically sampled transform (132_2,2) on the basis of the second segment (128_2,2) of the second set (128_2,1) of bins; and

wherein the time domain aliasing reduction stage (106) is configured to perform a weighted combination of the first set (110_1 ,1) of subband samples and the third set (110_2,1) of subband samples, to obtain a first aliasing reduced subband representation (112_1 ) of the audio signal, wherein the time domain aliasing reduction stage (106) is configured to perform a weighted combination of the second set (110_1,2) of subband samples and the fourth set (110_2,2) of subband samples, to obtain a second aliasing reduced subband representation (112_2) of the audio signal.

The audio processor (100) according to one of the claims 1 to 4, wherein the 5.

cascaded lapped critically sampled transform stage (104) is configured to segment a set (124_1) of bins obtained on the basis of the first block (108_1) of samples using at least two window functions, and to obtain at least two segmented sets (128_1,1;128_1,2) of subband samples based on the segmented set of bins corresponding to the first block (108_1) of samples;

wherein the cascaded lapped critically sampled transform stage (104) is configured to segment a set (124_2) of bins obtained on the basis of the second block (108_2) of samples using the at least two window functions, and to obtain at least two segmented sets (128_2,1;128_2,2) of subband samples based on the segmented set of bins corresponding to the second block (108_2) of samples; and

wherein the at least two window functions comprise different window width.

6. The audio processor (100) according to one of the claims 1 to 5, wherein the cascaded lapped critically sampled transform stage (104) is configured to segment a set (124_1) of bins obtained on the basis of the first block (108_1) of samples using at least two window functions, and to obtain at least two segmented sets (128_1 , 1 ; 128_1 ,2) of subband samples based on the segmented set of bins corresponding to the first block (108_1) of samples;

wherein the cascaded lapped critically sampled transform stage (104) is configured to segment a set (124_2) of bins obtained on the basis of the second block (108_2) of samples using the at least two window functions, and to obtain at least two sets (128_2,1;128_2,2) of subband samples based on the segmented set of bins corresponding to the second block (108_2) of samples; and

wherein filter slopes of the window functions corresponding to adjacent sets of subband samples are symmetric.

7. The audio processor (100) according to one of the claims 1 to 6, wherein the cascaded lapped critically sampled transform stage (104) is configured to segment the samples of the audio signal into the first block (108_1) of samples and the second block (108_2) of samples using a first window function;

wherein the lapped critically sampled transform stage (104) is configured to segment a set (124_1) of bins obtained on the basis of the first block (108_1) of samples and a set (124_2) of bins obtained on the basis of the second block (108_2) of samples using a second window function, to obtain the corresponding subband samples; and

wherein the first window function and the second window function comprise different window width.

8. The audio processor (100) according to one of the claims 1 to 6, wherein the cascaded lapped critically sampled transform stage (104) is configured to segment the samples of the audio signal into the first block (108_1) of samples and the second block (108_2) of samples using a first window function;

wherein the cascaded lapped critically sampled transform stage (104) is configured to segment a set (124_1) of bins obtained on the basis of the first block (108_1) of samples and a set (124-2) of bins obtained on the basis of the second block (108-2) of samples using a second window function, to obtain the corresponding subband samples; and

wherein a window width of the first window function and a window width of the second window function are different from each other, wherein the window width of the first window function and the window width of the second window function differ from each other by a factor different from a power of two.

9. The audio processor (100) according to one of the claims 1 to 8, wherein the time domain aliasing reduction stage (106) is configured to perform the weighted combination of two corresponding sets of subband samples according to the following equation

to obtain the aliasing reduced subband representation of the audio signal, wherein yv,i(m) is a first aliasing reduced subband representation of the audio signal, yv,I-1(N-1- m) is a second aliasing reduced subband representation of the audio signal,
y l ) a set of subband samples on the basis of the second block of samples of the audio signal, is a set of subband samples on the basis of the first block of

samples of the audio signal, av(m) is..., bv(m) is..., c»(m) is... and dv(m) is....

10. An audio processor (200) for processing a subband representation of an audio signal to obtain the audio signal (102), the audio processor (200) comprising:

an inverse time domain aliasing reduction stage (202) configured to perform a weighted combination of two corresponding aliasing reduced subband representations of the audio signal (102), to obtain an aliased subband

representation, wherein the aliased subband representation is a set (110_1,1) of subband samples; and

a cascaded inverse lapped critically sampled transform stage (204) configured to perform a cascaded inverse lapped critically sampled transform on the set (110_1,1) of subband samples, to obtain a set (206_1,1) of samples associated with a block of samples of the audio signal (102).

11. The audio processor (200) according to claim 10, wherein the cascaded inverse lapped critically sampled transform stage (204) comprises a first inverse lapped critically sampled transform stage (208) configured to perform an inverse lapped critically sampled transform on the set (110_1,1) of subband samples, to obtain a set of bins (128_1,1) associated with a given subband of the audio signal; and

a first overlap and add stage (210) configured to perform a concatenation of sets of bins associated with a plurality of subbands of the audio signal, which comprises a weighted combination of the set (128_1,1) of bins associated with the given subband of the audio signal (102) with a set (128_1,2) of bins associated with another subband of the audio signal (102), to obtain a set (124_1) of bins associated with a block of samples of the audio signal (102).

12. The audio processor (200) according to claim 11, wherein the cascaded inverse lapped critically sampled transform stage (204) comprises a second inverse lapped critically sampled transform stage (212) configured to perform an inverse lapped critically sampled transform on the set (124_1) of bins associated with the block of samples of the audio signal (102), to obtain a set of samples associated with the block of samples of the audio signal (102).

13. The audio processor (200) according to claim 12, wherein the cascaded inverse lapped critically sampled transform stage (204) comprises a second overlap and add stage (214) configured to overlap and add the set (206_1,1) of samples associated with the block of samples of the audio signal (102) and another set (206_2,1) of samples associated with another block of samples of the audio signal (102), the block of samples and the another block of samples of the audio signal (102) partially overlapping, to obtain the audio signal (102).

14. The audio processor (200) according to one of the claims 10 to 13, wherein the inverse time domain aliasing reduction stage (202) is configured to perform the weighted combination of the two corresponding aliasing reduced subband representations of the audio signal (102) based on the following equation

to obtain the aliased subband representation, wherein yv,i(m) is a first aliasing reduced subband representation of the audio signal, yv,I-1(N-1-m) is a second aliasing reduced subband representation of the audio signal,
is a set of subband samples on the basis of the second block of samples of the audio signal,
is a set of subband samples on the basis of the first block of samples of the audio signal, av(m) is..., bv(m) is..., Cy(m) is... and dv(m) is....

15. An audio encoder, comprising;

an audio processor (100) according to one of the claims 1 to 9;

an encoder configured to encode the aliasing reduced subband representation of the audio signal, to obtain an encoded aliasing reduced subband representation of the audio signal; and

a bitstream former configured to form a bitstream from the encoded aliasing reduced subband representation of the audio signal.

16. An audio decoder, comprising;

a bitstream parser configured to parse the bitstream, to obtain the encoded aliasing reduced subband representation;

a decoder configured to decode the encoded aliasing reduced subband representation, to obtain the aliasing reduced subband representation of the audio signal; and

an audio processor (200) according to one of the claims 10 to 14.

17. An audio analyzer, comprising:

an audio processor (100) according to one of the claims 1 to 9; and

an information extractor, configured to analyze the aliasing reduced subband representation, to provide an information describing the audio signal.

A method (300) for processing an audio signal to obtain a subband representation of 18.

the audio signal, the method comprising:

performing (302) a cascaded lapped critically sampled transform on at least two partially overlapping blocks of samples of the audio signal, to obtain a set of subband samples on the basis of a first block of samples of the audio signal, and to obtain a corresponding set of subband samples on the basis of a second block of samples of the audio signal; and

performing (304) a weighted combination of two corresponding sets of subband samples, one obtained on the basis of the first block of samples of the audio signal and one obtained on the basis on the second block of samples of the audio signal, to obtain an aliasing reduced subband representation of the audio signal.

A method (400) for processing a subband representation of an audio signal to obtain 19.

the audio signal, the method comprising:

Performing (402) a weighted combination of two corresponding aliasing reduced subband representations of the audio signal, to obtain an aliased subband representation, wherein the aliased subband representation is a set of subband samples; and

performing (404) a cascaded inverse lapped critically sampled transform on the set of subband samples, to obtain a set of samples associated with a block of samples of the audio signal.

20. A computer program for performing a method according to one of the claims 18 and

19.