AUDIO QUANTIZER, AUDIO DEQUANTIZER, AND RELATED METHODS - Patent application

Abstract

an audio quantizer for quantizing a plurality of audio information items, the audio quantizer comprising: a first-stage vector quantizer (141, 143) for quantizing the plurality of audio information items to determine a first-stage vector quantization result and a plurality of intermediate quantization items corresponding to the first-stage vector quantization result; a residual item determiner (142) for calculating a plurality of residual items from the plurality of intermediate quantization items and the plurality of audio information items; and a second-stage vector quantizer (145) for quantizing the plurality of residual items to obtain a second-stage vector quantization result, the first-stage vector quantization result and the second-stage vector quantization result being quantized representations of the plurality of audio information items.

Description

[Technical field]

[0001] Specifications and Preferred Embodiments The present invention relates to audio signal processing and can be applied, for example, in MDCT stereo processing of IVAS. Furthermore, the present invention can be applied in joint coding of stereo spectral noise shaping parameters. Spectral noise shaping shapes the quantization noise in the frequency domain so that it is minimally perceived by the human ear and thus the perceptual quality of the decoded output signal can be maximized. Spectral noise shaping is a technique used in most state-of-the-art transform-based audio codecs. [Background technology]

[0002] Advanced Audio Coding (AAC) In this technique [1][2], the MDCT spectrum is partitioned into several non-uniform scale factor bands. For example, at 48 kHz, the MDCT has 1024 coefficients and is partitioned into 49 scale factor bands. In each band, a scale factor is used to scale the MDCT coefficients of that band. Then, a scalar quantizer with a constant step size is used to quantize the scaled MDCT coefficients. At the decoder side, inverse scaling is performed in each band, which shapes the quantization noise introduced by the scalar quantizer. The 49 scale factors are coded into the bitstream as side information. Due to the relatively large number of scale factors and the required high precision, a significant number of bits are usually required to code these scale factors, which can be problematic at low bit rates and / or low delays.

[0003] MDCT-based TCX In the transform-based audio codec, MDCT-based TCX, used in the MPEG-D USAC [3] and 3GPP EVS [4] standards, spectral noise shaping is performed with the help of an LPC-based perceptual filter, similar to the one used in recent ACELP-based speech codecs (e.g. AMR-WB). In this approach, first, a set of 16 linear prediction coefficients (LPCs) is estimated for the pre-emphasized input signal. The LPCs are then weighted and quantized. The frequency response of the weighted and quantized LPCs is then calculated in 64 uniformly spaced bands. The MDCT coefficients are then scaled in each band using the calculated frequency response. The scaled MDCT coefficients are then quantized using a scalar quantizer with a step size controlled by a global gain. At the decoder, inverse scaling is performed in all 64 bands, which shapes the quantization noise introduced by the scalar quantizer. This approach has a clear advantage over the AAC approach, namely, it requires the coding of only 16 (LPC)+1 (global gain) parameters as side information (as opposed to 49 parameters in AAC). Moreover, the 16 LPCs can be efficiently coded with fewer bits by using an LSF representation and a vector quantizer. As a result, the MDCT-based TCX approach does not require as many side information bits as the AAC approach, which can make a significant difference in low bit rates and / or low delays.

[0004] Improved MDCT-based TCX (Psychoacoustic LPC) An improved MDCT-based TCX system is presented in [5]. In this new approach, the autocorrelation (to estimate the LPC) is no longer performed in the time domain, but instead is calculated in the MDCT domain using the inverse transform of the MDCT coefficient energies. This allows the use of a non-uniform frequency scale by simply grouping the MDCT coefficients into 64 non-uniform bands and calculating the energy of each band. This also reduces the complexity required to calculate the autocorrelation.

[0005] Novel Spectral Noise Shaping (SNS) In an improved technique for spectral noise shaping described in [6] and implemented in low-complexity communication codecs (LC3 / LC3plus), low bit rates without substantial loss of quality can be obtained by scaling with a larger number of scale factors at the encoder side and by downsampling the scale parameters at the encoder side to a second set of 16 scale parameters (SNS parameters). Thus, despite the low bit rate side information obtained on the one hand, high quality spectral processing of the audio signal spectrum is obtained due to the fine scaling on the other hand.

[0006] Stereo Linear Prediction (SLP) In the paper described in [7], a set of linear prediction coefficients is calculated by considering not only interframe prediction, but also per-channel prediction. The calculated two-dimensional set of coefficients is then quantized and coded using techniques similar to those for single-channel LP, but without considering quantization of the residual in the context of the paper. However, the described implementation involves high delay and significant complexity, and is therefore rather unsuitable for real-time applications requiring low delay, such as communication systems. In stereo systems, such as the MDCT-based system described in [8], pre-processing of the discrete LR channel signals is performed to scale the spectrum using frequency-domain noise shaping into a "whitened domain". Joint stereo processing is then performed to quantize and code the whitened spectrum in an optimal manner. The scaling parameters for the spectral noise shaping techniques described above are quantized and coded independently for each channel, resulting in double the bit rate of the side information that needs to be sent to the decoder via the bitstream. Summary of the Invention [Problem to be solved by the invention]

[0007] It is an object of the present invention to provide an improved or more efficient coding / decoding concept. [Means for solving the problem]

[0008] The object is to provide an audio quantizer according to claim 1, An audio inverse quantizer according to claim 15, Claim 29 The quantization method according to claim 30 or claim 1. 31 This is achieved by the computer program described in The invention is based on the discovery that bitrate savings can be obtained when the L, R signals, or in general two or more channels of a multi-channel signal, are correlated. In such cases, the extracted parameters for both channels are fairly similar. A joint quantization coding of the parameters is therefore applied, which results in significant bitrate savings. This bitrate savings can be used in several different directions. One direction can be to spend the saved bitrate on the coding of the core signal, such that the overall perceptual quality of the stereo or multi-channel signal is improved. Another direction is to reach a lower overall bitrate, when the coding of the core signal, and therefore the overall perceptual quality, is not improved and remains the same.

[0009] In a preferred embodiment, according to a first aspect, an audio encoder includes a scale parameter calculator for calculating a first group of jointly encoded scale parameters and a second group of jointly encoded scale parameters for a first set of scale parameters for a first channel of the multi-channel audio signal and a second set of scale parameters for a second channel of the multi-channel audio signal. The audio encoder further includes a signal processor for applying the first set of scale parameters to the first channel and applying the second set of scale parameters to the second channel of the multi-channel audio signal. The signal processor further derives multi-channel audio data from the first and second channel data obtained by application of the first and second sets of scale parameters, respectively. The audio encoder further includes an encoded signal former for using the multi-channel audio data, information about the first group of jointly encoded scale parameters and information about the second group of jointly encoded scale parameters to obtain an encoded multi-channel audio signal.

[0010] Preferably, the scale parameter calculator is configured to be adaptive, such that for each frame or subframe of the multi-channel audio signal a decision is made whether joint coding of the scale parameters or separate coding of the scale parameters should be performed. In a further embodiment, this decision is based on a similarity analysis between the channels of the multi-channel audio signal under consideration. In particular, the similarity analysis is made by calculating the energies of the jointly coded parameters, and in particular the energies of one set of scale parameters from the first group and the second group of the jointly coded scale parameters. In particular, the scale parameter calculator calculates the first group as the sum of the corresponding first and second scale parameters and the second group as the difference between the corresponding first and second scale parameters. In particular, the second group, and preferably the scale parameter representing the difference, is used for the similarity determination to determine whether joint coding of the scale parameters or separate coding of the scale parameters should be performed. This situation can be signaled via a stereo or multi-channel flag.

[0011] Moreover, it is preferred to specifically quantize the scale parameters in a two-stage quantization process. A first stage vector quantizer determines a first stage vector quantization result and quantizes a plurality of scale parameters, or in general, audio information items, to determine a plurality of intermediate quantizer items corresponding to the first stage vector quantization result. Furthermore, the quantizer includes a residual item determiner for calculating a plurality of residual items from the plurality of intermediate quantization items and the plurality of audio information items. Furthermore, a second stage vector quantizer is provided for quantizing the plurality of residual items to obtain a second stage vector quantization result, where the first stage vector quantization result and the second stage vector quantization result together represent a quantized representation of the plurality of audio information items, which in one embodiment are scale parameters. In particular, the audio information items may be either jointly coded scale parameters or separately coded scale parameters. Furthermore, the other audio information items may be any audio information items that are useful for vector quantization. In particular, apart from scale parameters or scale factors as a particular audio information item, the other audio information items that are useful for vector quantization are spectral values ​​such as MDCT or FFT lines. Still further audio information items which may be vector quantized are time domain audio values, such as audio sampling values, or groups of time domain audio samples, or groups of spectral domain frequency lines, or LPC data, or other envelope data, whether it be a spectral or temporal envelope data representation.

[0012] In a preferred implementation, the residual item determiner calculates for each residual item the difference between a corresponding audio information item, such as a scale parameter, and a corresponding intermediate quantization item, such as a quantized scale parameter or scale factor. Furthermore, the residual item determiner is configured to amplify or weight the difference between the corresponding audio information item and the corresponding intermediate quantization item, such that the multiple residual items are larger than the corresponding difference, or to amplify or weight the multiple audio information items and / or the multiple intermediate quantization items before calculating the difference between the amplified items to obtain the residual item. This procedure may provide a useful control of the quantization error. In particular, when the second group of audio information items, such as different scale parameters, is very small, the residual item is typically very small, which is typically the case when the first and second channels are correlated with each other as the joint quantization was determined. Thus, when the residual item is amplified, the result of the quantization will contain more values ​​that are not quantized to 0 compared to the case in which this amplification was not performed. Therefore, amplification at the encoder or quantization side may be useful.

[0013] This is especially true when quantization of the jointly coded second group of scale parameters, such as the difference scale parameters, is performed, as is the case in another preferred embodiment. Due to the fact that these side scale parameters are small in any case, a situation may arise where, without amplification, most of the different scale parameters are quantized to 0 in any case. Therefore, in order to avoid this situation, which may result in a loss of the stereo impression and therefore a loss of the psychoacoustic quality, an amplification is performed such that only a small amount of the side scale parameters are quantized to 0 or most of the side scale parameters are not quantized to 0. This naturally reduces the bitrate savings. However, due to this fact, the quantized residual data items are in any case small, i.e. resulting in quantization indices representing small values, and the increase in bitrate is not too high, since the quantization indices for small values ​​are coded more efficiently than the quantization indices for higher values. This can be further strengthened by additionally performing an entropy coding operation that gives even more preference to small quantization indices for the bitrate than higher quantization indices.

[0014] In another preferred embodiment, the first stage vector quantizer is a vector quantizer with a specific codebook, and the second stage vector quantizer is an algebraic vector quantizer that provides as quantization indexes the codebook number, the vector index in the base codebook, and the Voronoi index. Preferably, both the vector quantizer and the algebraic vector quantizer are configured to perform split-level vector quantization, where both quantizers have the same split-level procedure. Furthermore, the first and second stage vector quantizers are configured such that the number of bits, and therefore the precision, of the first stage vector quantizer result is greater than the number of bits or precision of the second stage vector quantizer result, or the number of bits, and therefore the precision, of the first stage vector quantizer result is different from the number of bits or precision of the second stage vector quantizer result. In other embodiments, the first stage vector quantizer has a fixed bitrate and the second stage vector quantizer has a variable bitrate. Thus, in general, the characteristics of the first stage vector quantizer and the second stage vector quantizer are different from each other. In a preferred embodiment of an audio decoder for decoding an encoded audio signal according to the first aspect, the audio decoder comprises a scale parameter decoder for decoding information about the jointly coded scale parameters, further comprising a signal processor, where the scale parameter decoder is configured to combine the jointly coded scale parameters of the first group and the jointly coded scale parameters of the second group using different combination rules to obtain scale parameters for the first set of scale parameters and scale parameters for the second set of scale parameters, which are subsequently used by the signal processor.

[0015] According to a further aspect of the present invention, there is provided an audio inverse quantizer comprising a first stage vector inverse quantizer, a second stage vector inverse quantizer and a combiner for combining a plurality of intermediate quantizer information items obtained by the first stage vector inverse quantizer and a plurality of residual items obtained from the second stage vector inverse quantizer to obtain a plurality of dequantized audio information items. The first aspect of the joint scale parameter coding may be combined with the second aspect of the two-stage vector quantization, while the latter aspect may be applied to separate coding scale parameters, such as a scale parameter for the left channel and a scale parameter for the right channel, or to the mid-scale parameter as another type of audio information item. Thus, the second aspect of the two-stage vector quantization may be applied independently of the first aspect or together with the first aspect.

[0016] Subsequently, preferred embodiments of the present invention are summarized. In stereo systems where transform-based (MDCT) coding is used, the scaling parameters extracted from any of the techniques described in the introduction for performing frequency-domain noise shaping at the encoder side need to be quantized and coded for inclusion as side information in the bitstream. At the decoder side, the scaling parameters are then decoded and used to scale the spectrum of each channel to minimize the perceived quantization noise. Independent coding of the spectral noise shaping parameters for the two channels, ie, the left and right channels, may be applied. The spectral noise shaping scaling parameters are adaptively coded independently or jointly depending on the degree of correlation between the two channels. · Mid / side representation of the scaling parameters is calculated The energy of the side parameters is calculated. Depending on the energy, which indicates the degree of correlation between the two signals, the parameters are coded as follows: Independently - for each channel, as in the current approach, e.g. using two-stage vector quantization (VQ) ·joint- o The mid-vectors are coded, for example, using two-stage vector quantization. The side-vectors are coded using a coarser quantization scheme, for example, by assuming that the first-stage VQ output contains quantized values ​​of zero and applying only the second-stage quantization, for example, an algebraic vector quantizer (AVQ). o One additional bit is used to signal whether the quantized side vector is zero or not One additional bit is sent to the decoder to signal whether the two channels are jointly or independently coded.

[0017] In Fig. 24, an implementation of an MDCT stereo based encoder is shown, as described in detail in [8]. An integral part of the stereo system described in [8] is that stereo processing is performed on the "whitened" spectrum. Thus, each channel undergoes a pre-processing, where for each frame, after windowing, the time domain blocks are transformed to the MDCT domain, and then temporal noise shaping (TNS) is adaptively applied, either before or after the spectral noise shaping (SNS) depending on the signal characteristics. After the spectral noise shaping, a joint stereo processing, i.e. an adaptive band-wise MS, L / R decision, is performed to efficiently quantize and code the whitened spectral coefficients. As a next step, a stereo intelligent gap filling (IGF) analysis is performed and the respective information bits are written to the bitstream. Finally, the processed coefficients are quantized and coded. The same reference numbers as in Fig. 1 are given. The calculation and processing of the scaling factors is performed in the block SNS between the two TNS blocks in Fig. 24. The block WINDOWS indicates the windowing operation. The block MCLT stands for the Modified Complex Lapped Transform. Block MDCT stands for Modified Discrete Cosine Transform. Block Power Spectrum stands for the calculation of the power spectrum. Block, Block Switching Decision, stands for the analysis of the input signal to determine the block length used for windowing. Block TNS stands for Temporal Noise Shaping, this feature is performed either before or after the spectral scaling in block SNS.

[0018] In the MDCT stereo codec implementation described in [7], pre-processing of the discrete LR channels is performed at the encoder side to scale the spectrum using frequency-domain noise shaping into a "whitened domain". Joint stereo processing is then performed to quantize and code the whitened spectrum in an optimal way. At the decoder side, the encoded signal is decoded and inverse quantization and inverse stereo processing is performed as shown in Fig. 25 and described in [8]. The spectrum of each channel is then "dewhitened" by the spectral noise shaping parameters retrieved from the bitstream. The same reference numbers as in Fig. 1 are used. The decoding and processing of the scale factors is performed in block 220 in Fig. 25. The blocks shown in the figure relate to the blocks in the encoder in Fig. 24 and typically perform the corresponding inverse operations. The block "Window and OLA" performs a synthesis windowing operation and a subsequent overlap and add operation to obtain the time domain output signals L and R.

[0019] Here, the frequency domain noise shaping (FDNS) applied in the system in [8] is replaced by SNS as described in [6]. The block diagram of the processing path of SNS is shown in the block diagrams of Figures 1 and 2 for the encoder and decoder, respectively. Preferably, a low bit rate without substantial loss of quality can be obtained by scaling with a larger number of scale factors at the encoder side and by downsampling the scale parameters at the encoder side to a second set of scale parameters or scale factors, where the scale parameters in the second set that are then encoded and transmitted or stored via the output interface are less than the first number of scale parameters. Thus, on the encoder side, fine scaling is obtained on the one hand and a low bit rate is obtained on the other hand. At the decoder side, the small number of transmitted scale factors are decoded by a scale factor decoder to obtain a first set of scale factors, where the number of scale factors or scale parameters in the first set is greater than the number of scale factors or scale parameters in the second set, and then fine scaling using a larger number of scale parameters is performed at the decoder side in the spectral processor to obtain a finely scaled spectral representation again. Thus, on the one hand a low bit rate is obtained, but on the other hand a high quality spectral processing of the audio signal spectrum is obtained.

[0020] Spectral noise shaping as performed in the preferred embodiment is implemented using only extremely low bit rates. Therefore, this spectral noise shaping can be an essential tool even in low bit rate transform-based audio codecs. Spectral noise shaping shapes the quantization noise in the frequency domain so that the quantization noise is minimally perceived by the human ear and thus the perceptual quality of the decoded output signal can be maximized. A preferred embodiment relies on spectral parameters calculated from amplitude-related measures, such as the energy of the spectral representation. In particular, the energy per band, or generally, the amplitude-related measures per band, are calculated as the basis for the scale parameters, where the bandwidth used in calculating the amplitude-related measures per band increases from lower bands to higher bands in order to come as close as possible to the characteristics of human hearing. Preferably, the division of the spectral representation into bands is performed according to the well-known Bark scale.

[0021] In a further embodiment, linear domain scale parameters are calculated, in particular for a first set of scale parameters having a large number of scale parameters, which are transformed into a logarithmic domain. The logarithmic domain is generally a domain in which small values ​​are expanded and high values ​​are compressed. A downsampling or decimation operation of the scale parameters is then performed in the logarithmic domain, which may be the logarithmic domain base 10 or the logarithmic domain base 2, where the latter is preferred for implementation purposes. A second set of scale factors is then calculated in the logarithmic domain, and preferably a vector quantization of the second set of scale factors is performed, where the scale factors are in the logarithmic domain. Thus, the result of the vector quantization represents the logarithmic domain scale parameters. The second set of scale factors or scale parameters has, for example, 1 / 2, even 1 / 3, and even more preferably 1 / 4 the number of scale factors of the first set. The quantized few scale parameters in the second set of scale parameters are then captured in a bitstream and then transmitted from the encoder side to the decoder side or stored as an encoded audio signal together with a quantized spectrum also processed using these parameters, where this processing further includes quantization using a global gain. However, preferably, the encoder again derives a set of linear domain scale factors, which is a third set of scale factors, from these quantized logarithmic domain second scale factors, where the number of scale factors in the third set of scale factors is greater than the second number and preferably even equal to the first number of scale factors in the first set of first scale factors. Then, at the encoder side, these interpolated scale factors are used to process the spectral representation, where the processed spectral representation is finally quantized and entropy coded in some way, such as Huffman coding, arithmetic coding, or vector quantization-based coding.

[0022] In a decoder receiving an encoded signal having a small number of spectral parameters together with the encoded representation of the spectral representation, the small number of scale parameters is interpolated to a large number of scale parameters, i.e. to obtain a first set of scale parameters, where the number of scale parameters of the scale factors or of the scale factors of the second set of scale parameters is smaller than the number of scale parameters of the first set, i.e. the set calculated by the scale factor / parameter decoder. A spectral processor located in the device for decoding the encoded audio signal then processes the decoded spectral representation using this first set of scale parameters to obtain a scaled spectral representation. A converter for converting the scaled spectral representation then operates to finally obtain a decoded audio signal, preferably in the time domain.

[0023] Further embodiments provide additional advantages, which are described below. In a preferred embodiment, the spectral noise shaping is performed with the help of 16 scaling parameters similar to the scale factors used in [6] or [8] or [1]. These parameters are obtained in the encoder by first calculating the energy of the MDCT spectrum in 64 non-uniform bands (similar to the 64 non-uniform bands of prior art 3), then applying some processing (smoothing, pre-emphasis, noise floor, logarithmic transformation) to the 64 energies, and then downsampling the 64 processed energies by a factor of 4 to obtain 16 parameters, which are finally normalized and scaled. These 16 parameters are then quantized using vector quantization (using vector quantization similar to that used in prior art 2 / 3). The quantized parameters are then interpolated to obtain 64 interpolated scaling parameters. These 64 scaling parameters are then used to directly shape the MDCT spectrum in the 64 non-uniform bands. As in prior arts 2 and 3, the scaled MDCT coefficients are then quantized using a scalar quantizer with the step size controlled by the global gain.

[0024] In a further embodiment, the information regarding the jointly coded scale parameters for one of the two groups, such as the second group preferably related to the side scale parameters, does not include quantization indices or other quantization bits, but only includes information such as a flag or a single bit indicating that the scale parameters for the second group are all zero for a portion or frame of the audio signal. This information is determined by the encoder by analysis or other means and used by the decoder to synthesize the second group of scale parameters based on this information, such as by generating zero scale parameters for the time portion or frame of the audio signal, or is used by the decoder to compute the first and second sets of scale parameters using only the first group of jointly coded scale parameters. In a further embodiment, the second group of jointly encoded scale parameters is quantized using only the second quantization stage of the two-stage quantizer, preferably a variable-rate quantizer stage. In this case, it is assumed that the first stage results in all-zero quantized values, and therefore only the second stage is effective. In yet a further embodiment, only the first quantization stage of the two-stage quantizer, preferably a fixed-rate quantization stage, is applied, and the second stage is not used at all for a time portion or frame of the audio signal. This case corresponds to a situation where all residual terms are assumed to be zero or smaller than the minimum or first quantization step size of the second quantization stage. Subsequently, preferred embodiments of the present invention will be described with reference to the accompanying drawings. [Brief description of the drawings]

[0025] [Figure 1] FIG. 2 shows a decoder according to a first embodiment; [Diagram 2] FIG. 2 shows an encoder according to a first embodiment; [Figure 3a] FIG. 2 illustrates another encoder according to a first embodiment. [Figure 3b]FIG. 2 illustrates another implementation of an encoder according to the first aspect. [Figure 4a] FIG. 4 shows a further embodiment of a decoder according to the first aspect; [Figure 4b] FIG. 2 illustrates another embodiment of a decoder. [Diagram 5] FIG. 2 illustrates a further embodiment of an encoder. [Figure 6] FIG. 2 illustrates a further embodiment of an encoder. [Figure 7a] FIG. 2 shows a preferred implementation of a vector quantizer according to the first or second aspect. [Figure 7b] FIG. 2 illustrates a further quantizer according to the first or second embodiment. [Figure 8a] FIG. 2 shows a decoder according to a first aspect of the invention; [Figure 8b] FIG. 2 shows an encoder according to a first aspect of the present invention; [Figure 9a] FIG. 2 shows an encoder according to a second aspect of the present invention. [Figure 9b] FIG. 3 shows a decoder according to a second aspect of the invention. [Figure 10] FIG. 2 shows a preferred implementation of a decoder according to the first or second aspect. [Figure 11] 1 is a block diagram of an apparatus for encoding an audio signal; [Figure 12] 2 is a schematic representation of a preferred implementation of the scale factor calculator of FIG. 1; [Figure 13] 2 is a schematic representation of a preferred implementation of the downsampler of FIG. 1; [Figure 14] FIG. 5 is a schematic representation of the scale factor encoder. [Figure 15] FIG. 2 is a schematic diagram of the spectral processor of FIG. 1; [Figure 16] 1 is a schematic representation of an encoder on the one hand and a decoder on the other hand, implementing Spectral Noise Shaping (SNS). [Figure 17]A more detailed representation on the one hand, the encoder side and on the other hand, where temporal noise shaping (TNS) is performed together with spectral noise shaping (SNS). [Figure 18] 1 is a block diagram of an apparatus for decoding an encoded audio signal; [Figure 19] FIG. 9 is a schematic diagram showing details of the scale factor decoder, spectral processor and spectral decoder of FIG. [Figure 20] FIG. 2 shows a subdivision of the spectrum into 64 bands. [Figure 21] FIG. 2 is a schematic diagram of a downsampling operation on the one hand and an interpolation operation on the other hand; [Figure 22a] FIG. 2 illustrates a time-domain audio signal with overlapping frames. [Figure 22b] FIG. 2 illustrates an implementation of the converter of FIG. 1. [Figure 22c] FIG. 9 is a schematic diagram of the converter of FIG. 8. [Diagram 23] 1 is a histogram comparing different inventive procedures. [Figure 24] FIG. 2 illustrates an embodiment of an encoder. [Diagram 25] FIG. 2 illustrates an embodiment of a decoder. DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS

[0026] Fig. 8 shows an audio decoder for decoding an encoded audio signal comprising multi-channel audio data comprising data for two or more audio channels and information on jointly encoded scale parameters. The decoder comprises a scale parameter decoder 220 and signal processors 210, 212, 213 shown as a single item in Fig. 8a. The scale parameter decoder 220 receives information on the jointly encoded first and second groups of scale parameters, where preferably the first group of scale parameters are mid-scale parameters and the second group of scale parameters are side-scale parameters. Preferably, the signal processor receives a first channel representation of the multi-channel audio data and a second channel representation of the multi-channel audio data, applies a first set of scale parameters to the first channel representation derived from the multi-channel audio data and applies a second set of scale parameters to the second channel representation derived from the multi-channel audio data to obtain the first and second channels of a decoded audio signal at the output of blocks 210, 212, 213 of Fig. 8a. Preferably, the joint coded scale parameters include information on a first group of joint coded scale parameters, such as mid-scale parameters, and information on a second group of joint coded scale parameters, such as side-scale parameters. Furthermore, the scale parameter decoder 220 is configured to combine the joint coded scale parameters of the first group and the joint coded scale parameters of the second group using a first combining rule to obtain scale parameters of the first set of scale parameters, and to combine both the same joint coded scale parameters of the first group and the second group using a second combining rule different from the first combining rule to obtain scale parameters of the second set of scale parameters. Thus, the scale parameter decoder 220 applies two different combining rules.

[0027] In a preferred embodiment, the two different combination rules are a plus or additive combination rule on the one hand and a subtractive or differential combination rule on the other hand. However, in other embodiments, the first combination rule can be a multiplication combination rule and the second combination rule can be a quotient or division combination rule. Thus, all other pairs of combination rules are also useful depending on the representation of the corresponding scale parameters in the first and second groups or first and second sets of scale parameters.

[0028] Fig. 8b shows a corresponding audio encoder for encoding a multi-channel audio signal comprising two or more channels. The audio encoder comprises a scale parameter calculator 140, a signal processor 120 and an encoded signal former 1480, 1500. The scale parameter calculator 140 is configured for calculating a first group of jointly encoded scale parameters and a second group of jointly encoded scale parameters from a first set of scale parameters for a first channel of the multi-channel audio signal and a second set of scale parameters for a second channel of the multi-channel audio signal. Further, the signal processor is configured for applying the first set of scale parameters to the first channel of the multi-channel audio signal and applying the second set of scale parameters to the second channel of the multi-channel audio signal to derive encoded multi-channel audio data. Multi-channel audio data is derived from the scaled first and second channels, which multi-channel audio data is used by the coded signal formers 1480, 1500 together with information about the first and second groups of joint coding scale parameters to obtain a coded multi-channel audio signal at the output of block 1500 in FIG. 8b.

[0029] Figure 1 shows a further implementation of the decoder of Figure 8a. In detail, the bitstream is input to a signal processor 210, which typically performs entropy decoding and inverse quantization together with an intelligent gap filling procedure (IGF procedure) and inverse stereo processing of the scaled or whitened channels. The output of block 210 is the scaled or whitened decoded left and right, or in general several decoded channels, of a multi-channel signal. The bitstream includes side information bits for left and right scale parameters in case of separate coding and side information bits for scaled joint coding scale parameters, shown as M, S scale parameters in Figure 1. This data is introduced into a scale parameter or scale factor decoder 220 which produces at its output a decoded left scale factor and a decoded right scale factor which are then applied in the spectral shaping blocks 212, 230 to finally obtain left and right preferably MDCT spectra which can then be transformed into the time domain using a specific inverse MDCT operation.

[0030] A corresponding encoder-side implementation is given in Fig. 2. Fig. 2 starts with an MDCT spectrum with left and right channels input to a spectrum shaper 120a, whose output is input to a processor 120b that performs, for example, stereo processing, an intelligent gap-filling operation at the encoder side, and corresponding quantization and (entropy) coding operations. Thus, blocks 120a, 120b together represent the signal processor 120 of Fig. 8b. Furthermore, for the purpose of the calculation of the scale factors performed in the SNS (spectral noise shaping) scale factor calculation block 120b, an MDST spectrum is also provided, which is forwarded together with the MDCT spectrum to the power spectrum calculator 110a. Alternatively, the power spectrum calculator 110a can operate directly on the input signal without the MDCT or MDST spectrum procedure. Another way is to calculate the power spectrum from, for example, a DFT operation rather than an MDCT and MDST operation. Further, the scale factors are calculated by a scale parameter calculator 140, shown in Fig. 2 as a scale factor quantization coding block. In detail, the block 140 outputs either left and right separate coding scale factors or M and S joint coding scale factors depending on the similarity between the first and second channels. This is shown to the right of the block 140 in Fig. 2. Thus, in this implementation, the block 110b calculates the left and right scale factors, and then the block 140 decides whether the separate coding, i.e., the coding for the left and right scale factors, is better or worse than the joint coding scale factors, i.e., the coding of the M and S scale factors derived from the separate scale factors by two different combination rules, such as addition on the one hand and subtraction on the other hand. The result of block 140 are L,R or M,S side information bits which, together with the result of block 120b, are introduced into the output bitstream shown in FIG.

[0031] Fig. 3a shows a preferred implementation of the encoder of Fig. 2 or Fig. 8b. The first channel is input to a block 1100a for determining separate scale parameters for the first channel, i.e. channel L. Furthermore, the second channel is input to a block 1100b for determining separate scale parameters for the second channel, i.e. R. The scale parameters for the left channel and the scale parameters for the right channel are then downsampled by the downsampler 130a for the first channel and the downsampler 130b for the second channel, correspondingly. The result is the downsampled parameters for the left channel (DL) and the downsampled parameters for the right channel (DR). Both of these data DL and DR are then input to a joint scale parameter determiner 1200, which generates a first group of jointly coded scale parameters, such as mid or M scale parameters, and a second group of jointly coded scale parameters, such as side or S scale parameters. Both groups are input in corresponding vector quantizers 140a, 140b to obtain quantized values, which are then in a final entropy encoder 140c to be coded to obtain information about the jointly coded scale parameters. The entropy encoder 140c may be implemented to perform an arithmetic entropy coding algorithm or an entropy coding algorithm using one-dimensional or more than one-dimensional Huffman code tables.

[0032] Another implementation of the encoder is shown in Fig. 3b, where downsampling is not performed on separate scale parameters, such as on the left and right as shown at 130a, 130b in Fig. 3a. Instead, the order of the joint scale parameter determination and the subsequent downsampling operations by the corresponding downsamplers 130a, 130b is changed. Whether the implementation of Fig. 3a or that of Fig. 3b is used depends on the particular implementation, where the implementation of Fig. 3a is preferred, since the joint scale parameter determination 1200 is already performed on the downsampled scale parameters, i.e., the two different combination rules performed by the scale parameter calculator 140 are typically performed on a smaller number of inputs compared to the case of Fig. 3b.

[0033] FIG. 4a shows an implementation of a decoder for decoding an encoded audio signal having multi-channel audio data including data for two or more audio channels and information on jointly encoded scale parameters. However, the decoder in FIG. 4a is only a part of the overall decoder in FIG. 8a, since only a part of the signal processor, in particular the corresponding channel scalers 212a, 212b, is shown in FIG. 4a. With regard to the scale parameter decoder 220, this element includes an entropy decoder 2200 that reverses the procedure performed by the corresponding block 140c in FIG. 3a. Furthermore, the entropy decoder outputs quantized jointly encoded scale parameters, such as a quantized M scale parameter and a quantized S scale parameter. The corresponding groups of scale parameters are input to inverse quantizers 2202 and 2204 to obtain inverse quantized values ​​of M and S. These inverse quantized values ​​are then input to a separate scale parameter determiner 2206 that outputs left and right scale parameters, i.e. separate scale parameters. These corresponding scale parameters are input to the interpolators 222a, 222b to obtain the left interpolated scale parameter (IL) and the right interpolated scale parameter (IR). Both of these data are input to the channel scalers 212a, 212b, respectively. Furthermore, the channel scalers receive the first channel representation correspondingly, for example after the entire procedure performed by the block 210 in FIG. 1. Correspondingly, the channel scaler 212b also obtains its corresponding second channel representation output by the block 210 in FIG. 1. Then, a final channel scaling or "spectral shaping" as it is called in FIG. 1 is performed to obtain the left and right shaped spectral channels, shown as "MDCT spectrum" in FIG. 1. Then, a final frequency domain to time domain transform for each channel, shown at 240a, 240b, may be performed to finally obtain the decoded first channel and the decoded second channel of the multi-channel audio signal in the time domain representation.

[0034] In particular, the scale parameter decoder 220 shown in the left part of Fig. 4a may be included within the audio decoder as shown in Fig. 1 or as shown collectively in Fig. 4a, but may also be included as a local decoder within the encoder, as shown with respect to Fig. 5, which explicitly shows the local scale parameter decoder 220 at the output of the scale parameter encoder 140. Fig. 4b shows a further implementation in which the order of the interpolation and the scale parameter determination for determining the separate scale parameters is swapped with respect to Fig. 4a. In detail, the interpolation is performed on the joint coded scale parameters M and S using the interpolators 222a, 222b of Fig. 4b, and the interpolated joint coded scale parameters such as IM and IS are input to the separate scale parameter determiner 2206. The output of the block 2206 is then the upsampled scale parameters, i.e. the scale parameters for each of the 64 bands shown in Fig. 21 for example.

[0035] Fig. 5 shows a further preferred implementation of the encoder of Fig. 8b, Fig. 2 or Fig. 3a, Fig. 3b. Both the first channel and the second channel are introduced into an optional time domain to frequency domain transformer, such as 100a, 100b in Fig. 5. The spectral representations output by the blocks 100a, 100b are input to a channel scaler 120a, which scales the spectral representations for the left and right channels separately. Thus, the channel scaler 120a performs the spectral shaping operation shown in 120a in Fig. 2. The output of the channel scaler is input to the channel processor 120b in Fig. 5, and the processed channel output of the block 120b is input to the coded signal former 1480, 1500 to obtain the coded audio signal. Furthermore, for the purpose of determining separate or joint coding scale parameters, a similarity calculator 1400 is provided which receives as input the first and second channels directly in the time domain. Alternatively, the similarity calculator can receive the first and second channels at the output of the time domain to frequency domain converters 100a, 100b, i.e. in their spectral representation.

[0036] With respect to FIG. 6, it is outlined that the similarity between the two channels is calculated based on the second group of joint coding scale parameters, i.e., based on the side scale parameters, but it should be noted that this similarity can also be calculated based on the time domain or the spectral domain channels directly without the explicit calculation of the joint coding scale parameters. Alternatively, the similarity can also be determined based on the first group of joint coding scale parameters, i.e., based on the mid-scale parameters. In detail, it is determined that joint coding can be performed when the energy of the side scale parameters is lower than a threshold. Similarly, the energy of the mid-scale parameters in the frame can also be measured, and for example, a decision of joint coding can be made when the energy of the mid-scale parameters is greater than another threshold. Thus, many different methods can be implemented for determining the similarity between the first channel and the second channel to determine the joint coding of the scale parameters or the separate coding of the scale parameters. Nevertheless, it should be noted that the decision of joint or separate coding of the scale parameters does not necessarily have to be identical to the decision of joint stereo coding for the channels, i.e., whether the two channels are jointly coded using the mid / side representation or separately coded in the L, R representation. The decision of joint coding of the scale parameters is made independently of the decision of stereo processing for the actual channels, since the decision of any kind of stereo processing performed in block 120b in FIG. 2 is made after and subsequent to the scaling or shaping of the spectrum using the mid and side scale factors. In particular, as shown in FIG. 2, block 140 can decide on joint coding. Thus, the M and S scale factors can occur in this block, as indicated by the arrow in FIG. 2 pointing to block 140.In the case of application of the local scale parameter decoder 220 in the encoder of FIG. 5, the scale parameters actually used to shape the spectrum are the left scale parameter and the right scale parameter, which are nevertheless derived from the mid and side encoding and decoding scale parameters.

[0037] With reference to Fig. 5, a mode determiner 1402 is provided. The mode determiner 1402 receives the output of the similarity calculator 1400 and determines separate coding of the scale parameters when the channels are not sufficiently similar. However, when the channels are determined to be similar, joint coding of the scale parameters is determined by the block 1402 and information, whether separate coding or modified joint coding of the scale parameters is applied, is signaled by corresponding side information or flag 1403 shown in Fig. 5, which is provided from the block 1402 to the coding signal formers 1480, 1500. Furthermore, the encoder includes a scale parameter encoder 140 that receives the scale parameters for the first channel and the scale parameters for the second channel and separately codes or jointly codes the scale parameters as controlled by the mode determiner 1402. The scale parameter encoder 140 may, in one embodiment, output scale parameters for the first and second channels as indicated by the dashed lines, such that the channel scaler 120a performs scaling with the corresponding first and second channel scale parameters. However, it is preferred to apply a local scale parameter decoder 220 in the encoder, whereby channel scaling is performed with locally encoded and decoded scale parameters, such that the dequantized scale parameters are applied for channel scaling in the encoder. This has the advantage that exactly the same situation occurs in the channel scalers in the encoder and the decoder, at least with regard to the scale parameters used for channel scaling or spectral shaping.

[0038] Fig. 6 shows a further preferred embodiment of the invention for an audio encoder. An MDCT spectrum calculator 100 is provided, which may be, for example, a time domain to frequency domain transformer applying an MDCT algorithm. Furthermore, as shown in Fig. 2, a power spectrum calculator 110a is provided. Separate scale parameters are calculated by the corresponding calculator 1100, as well as by the addition block 1200a and the subtraction block 1200b, for the purpose of calculating joint coding scale parameters. Then, a frame-by-frame energy calculation is performed with the side parameters, i.e. the second group of joint coding scale parameters, for the purpose of determining similarity. In block 1406, a comparison with a threshold is performed, and this block, which is similar to the mode determiner 1402 for a frame in Fig. 5, outputs a mode flag or a stereo flag for the corresponding frame. Furthermore, the information is given to a controllable encoder that performs separate or joint coding in the current frame. For this purpose, the controllable encoder 140 receives the scale parameters calculated by block 1100, i.e. the separate scale parameters, and further receives the jointly encoded scale parameters, i.e. the scale parameters determined by blocks 1200a and 1200b.

[0039] Block 140 preferably generates a zero flag for a frame when block 140 determines that all side parameters of the frame are quantized to 0. This result occurs when the first and second channels are very close to each other and the differences between the channels, and therefore the differences between the scale factors, are such that these differences are smaller than the lowest quantization threshold applied by the quantizer included in block 140. Block 140 outputs information regarding the joint coding or separate coding scale parameters for the corresponding frame. Fig. 9a shows an audio quantizer for quantizing a plurality of audio information items. The audio quantizer includes first stage vector quantizers 141, 143 for quantizing a plurality of audio information items, such as scale factors or scale parameters or spectral values, to determine a first stage vector quantization result 146. Furthermore, blocks 141, 143 generate a plurality of intermediate quantization items corresponding to the first stage vector quantization result. The intermediate quantization items are, for example, values ​​associated with the first stage result. When the first stage result identifies a particular codebook having, for example, 16 particular (quantized) values, the intermediate quantization items are 16 values ​​associated with the codebook vector index, which is the first stage result 146. The intermediate quantization items and audio information items at the input to the first stage vector quantizers 141, 143 are input to a residual item determiner for calculating a plurality of residual items from the plurality of intermediate quantization items and the plurality of audio information items, for example by calculating the difference for each item between the original item and the quantized item. The residual items are input to a second stage vector quantizer 145 for quantizing the multiple residual items to obtain a second stage vector quantization result. The first stage vector quantization result at the output of blocks 141, 143 and the second stage result at the output of block 145 together then represent a quantized representation of the multiple audio information items, which is coded by an optional coding signal former 1480, 1500 which outputs a quantized audio information item, which in a preferred embodiment is not only quantized but also entropy coded.

[0040] A corresponding audio inverse quantizer is shown in Fig. 9b. The audio inverse quantizer comprises a first stage vector inverse quantizer 2220 for inverse quantizing the first stage quantization results contained in the quantized audio information items to obtain a plurality of intermediate quantized audio information items. Furthermore, a second stage vector inverse quantizer 2260 is provided, configured for inverse quantizing the second stage vector quantization results contained in the quantized audio information items to obtain a plurality of residual items. Both the intermediate items from the block 2220 and the residual items from the block 2260 are combined by a combiner 2240 for combining the plurality of intermediate quantized audio items and the plurality of residual items to obtain a plurality of dequantized audio information items. In detail, the intermediate quantization items at the output of the block 2220 may represent separate coded scale parameters, such as L and R, or a first group of joint coded scale parameters, e.g. M, and the residual items may represent joint coded side scale parameters, e.g. a second group of joint coded scale parameters, i.e.

[0041] FIG. 7a shows a preferred implementation of the first stage vector quantizer 141, 143 of FIG. 9a. In step 701, vector quantization of a first subset of scale parameters is performed to obtain a first quantization index. In step 702, vector quantization of a second subset of scale parameters is performed to obtain a second quantization index. Furthermore, vector quantization of a third subset of scale parameters is performed, as shown in block 703, to obtain a third quantization index, which is an optional index depending on the implementation. The procedure of FIG. 7a is applied when there is split-level quantization. Exemplarily, the audio input signal is separated into 64 bands as shown in FIG. 21. These 64 bands are downsampled to 16 bands / scale factors, so that the entire band is covered by 16 scale factors. These 16 scale factors are quantized by the first stage vector quantizer 141, 143 in the split-level mode shown in FIG. 7a. The first 8 scale factors of the 16 scale factors of Fig. 21 obtained by downsampling the original 64 scale factors are vector quantized by step 701 and thus represent a first subset of scale parameters. The remaining 8 scale parameters for the 8 upper bands represent a second subset of scale parameters that are vector quantized in step 702. Depending on the implementation, the separation of the scale parameters or the whole set of audio information items does not necessarily have to be done in exactly two subsets, but could also be in three or even more subsets.

[0042] Regardless of how many splits are performed, the indices for each level together represent the first stage result. As described with respect to Figure 14, these indices may be combined via an index combiner in Figure 14 to have a single first stage index. Alternatively, the first stage result may be composed of a first index, a second index, and potentially a third index, and possibly even more indexes, which are not combined but are entropy coded as is.

[0043] In addition to the corresponding indexes forming the first stage result, steps 701, 702, 703 also provide intermediate scale parameters that are used in block 704 to calculate residual scale parameters for the frame. Thus, step 705, for example performed by block 142 of Fig. 9a, results in residual scale parameters that are then processed by (algebraic) vector quantization performed by step 705 to generate a second stage result. Thus, a first stage result and a second stage result are generated for the first group of separate scale parameters L, separate scale parameters R and joint scale parameters M. However, as shown in Fig. 7b, the (algebraic) vector quantization of the second group of jointly coded scale parameters or side scale parameters is only performed by step 706, which in a preferred implementation is identical to step 705, again performed by block 142 of Fig. 9a.

[0044] In a further embodiment, the information regarding the jointly coded scale parameters for one of the two groups, such as the second group, preferably related to the side scale parameters, does not include a quantization index or other quantization bits, but only information such as a flag or a single bit indicating that the scale parameters for the second group are all zero for a portion or frame of the audio signal, or are all at a particular value, such as a small value. This information is determined by the encoder by analysis or other means, and is used by the decoder to synthesize the second group of scale parameters based on this information, such as by generating zero scale parameters for the time portion or frame of the audio signal, or by generating scale parameters of a particular value, or by generating small random scale parameters, e.g. all smaller than the minimum or first quantization stage, or by the decoder to calculate the first and second sets of scale parameters using only the first group of jointly coded scale parameters. Thus, instead of performing stage 705 in Fig. 7a, only an all-zero flag for the second group of jointly coded scale parameters is written as the second stage result. The calculation in block 704 may again be omitted and replaced by a determiner for determining whether the all-zero flag is to be activated and transmitted. This determiner may be controlled by a user input or bit rate information indicating skipping the coding of the entire S-parameters, or may actually perform an analysis of the residual terms. Thus, for frames with all-zero bits, the scale parameter decoder does not perform any combination and calculates the second set of scale parameters using only the first group of jointly coded scale parameters, such as by dividing the coded scale parameters of the first group by two or by weighting them using another predefined value.

[0045] In a further embodiment, the second group of jointly encoded scale parameters is quantized using only the second quantization stage of a two-stage quantizer, preferably a variable rate quantizer stage. In this case, the first stage is assumed to result in all-zero quantized values, and therefore only the second stage is effective. This case is shown in Figure 7b. In yet a further embodiment, only the first quantization stage, such as 701, 702, 703 of the two-stage quantizer in Fig. 7a, which is preferably a fixed-rate quantization stage, is applied, and the second stage 705 is not used at all for a time portion or frame of the audio signal. This case corresponds to a situation in which all residual terms are assumed to be zero or smaller than the minimum or first quantization step size of the second quantization stage. Then Fig. 7b, item 706 corresponds to items 701, 702, 703 in Fig. 7a, and item 704 may be omitted as well and replaced by a decider for deciding that only the first stage quantization is used or not. This decider may be controlled by user input or bit rate information, or may actually perform an analysis of the residual terms to determine that the residual terms are very small and the accuracy of the second group of joint coding scale parameters quantized by only a single stage is sufficient.

[0046] In a preferred implementation of the present invention, further illustrated in Fig. 14, the algebraic vector quantizer 145 further performs a split level calculation, preferably performing the same split level operation as performed by the vector quantizer. Thus, a subset of residual values ​​corresponds to a subset of scale parameters in terms of band number. In the case of having two split levels, i.e. for the first eight downsampled bands of Fig. 21, the algebraic vector quantizer 145 generates a first level result. Furthermore, the algebraic vector quantizer 145 generates a second level result for the top eight downsampled scale factors or scale parameters, or audio information items in general. Preferably, the algebraic vector quantizer 145 is implemented as an algebraic vector quantizer defined in section 5.2.3.1.6.9 of ETSI TS126 445 V13.2.0 (2016-08), referred to as reference (4), where the corresponding split multi-rate lattice vector quantization result is a codebook number, a vector index in a base codebook, and an 8-dimensional Voronoi index for each of the 8 items. However, when having only a single codebook, the codebook number can be avoided, and only the vector index in a base codebook and the corresponding n-dimensional Voronoi index are sufficient. Thus, for each level for the algebraic vector quantization result, these items, which are only item a, item b, and item c, or item b and item c, represent the second stage quantization result.

[0047] Reference is now made to FIG. 10 which illustrates a corresponding decoding operation consistent with the encoding of FIG. 7a, FIG. 7b or the encoding of FIG. 14 according to the first or second aspect of the invention or both aspects. In step 2221 of FIG. 10, the quantized mid-scale factors, i.e. the second group of jointly encoded scale factors, are retrieved. This is done when the stereo mode flag or item 1403 of FIG. 5 indicates a true value. Then, a first stage decoding 2223 and a second stage decoding 2261 are performed to re-perform the procedure performed by the encoder of FIG. 14, in particular by the algebraic vector quantizer 145 described with respect to FIG. 14 or with respect to FIG. 7a. In step 2225, the side scale factors are assumed to be all zero. In step 2261, it is checked whether the zero flag value actually results in a non-zero quantized scale factor for the frame. If the zero flag value indicates that there is a non-zero side scale factor for the frame, the quantized side scale factors are retrieved and decoded using the second stage decoding 2261 or by only performing block 706 of FIG. 7b. In block 2207, the jointly coded scale parameters are converted to separate coded scale parameters to subsequently output quantized left and right scale parameters, which may then be used for inverse scaling of the spectrum at the decoder. When the stereo mode flag value indicates a value of zero or when it is determined that separate coding was used in the frame, only the first stage decoding 2223 and the second stage decoding 2261 are performed for the left and right scale factors and no transformations such as block 2207 are required since the left and right scale factors are already in the separately coded representation. The process of efficiently coding and decoding the SNS scale factors required for scaling the spectrum before stereo processing at the encoder side and after inverse stereo processing at the decoder side is described below to illustrate a preferred implementation of the present invention as an exemplary pseudo code with comments.

[0048] Joint quantization and coding of scale factors JPEG0007641355000001.jpg138117 JPEG0007641355000002.jpg170120

[0049] Any kind of quantization may be used to represent the parameters, for example, uniform or non-uniform scalar quantization and entropy or arithmetic coding. In the described implementation, as can be seen from the algorithm description, a two-stage vector quantization scheme is implemented. 1st stage - 2 splits (8 dimensions each) using 5 bits each, so coded using 10 bits · Second stage - Algebraic Vector Quantization (AVQ), also two-split, with scaling of the residual, where the codebook indices are entropy coded, thus using a variable bit rate. Since the side signals for highly correlated channels can be considered small, it is sufficient to use only, for example, a reduced-scale second-stage AVQ to represent the corresponding SNS parameters. By skipping the first-stage VQ for these signals, significant complexity and bit savings for the coding of the SNS parameters can be achieved.

[0050] A pseudocode description of each stage of quantization performed is given below: First stage with 2-split vector quantization using 5 bits for each split. JPEG0007641355000003.jpg169116 JPEG0007641355000004.jpg128106

[0051] Second-stage algebraic vector quantization. JPEG0007641355000005.jpg109114 In an embodiment of the audio quantizer, the first stage vector quantizer (141, 143) is configured to output a first quantization index having a first number of bits from a first codebook lookup, and the second stage vector quantizer (145) is configured to output a second quantization index having a second number of bits from a second codebook lookup, the second number of bits being less or more than the first number of bits, the first number of bits being between 4 and 7 bits, and the second number of bits being between 3 and 6 bits.

[0052] The indexes output from the coding process are finally packed into a bitstream and sent to the decoder. The AVQ procedure disclosed above for the second stage is preferably implemented as outlined in EVS for high-rate LPC (subclause 5.3.3.2.1.3) in the MDCT-based TCX chapter. In particular, refinements are described in subclause 5.2.3.1.6.9 for the second stage algebraic vector quantizer used, described as 5.3.3.2.1.3.4 Algebraic Vector Quantizer, and the algebraic VQ used for quantization. In one embodiment, for each index, we have a set of codewords for the base codebook index and a set of codewords for the Voronoi index, all of which are entropy coded and therefore variable bitrate. Thus, the parameters of the AVQ in each subband j consist of the codebook number, the vector index in the base codebook, and the n-dimensional (e.g. 8-dimensional) Voronoi index. In an embodiment of the audio quantizer, an audio information item comprises, for a first frame of the multi-channel audio signal, a first plurality of scale parameters for a first channel of the multi-channel audio signal and a second plurality of scale parameters for a second channel of the multi-channel audio signal. The audio quantizer is configured to apply a first stage vector quantizer (141, 143) and a second stage vector quantizer (145) to the first plurality of scale parameters for the first channel of the first frame and the second plurality of scale parameters for the second channel of the first frame. The audio information item comprises, for a second frame of the multi-channel audio signal, a third plurality of mid-scale parameters and a fourth plurality of side-scale parameters. The audio quantizer is configured to apply a first stage vector quantizer (141, 143) and a second stage vector quantizer (145) to a third plurality of mid-scale parameters of the second frame, to apply the second stage vector quantizer (145) to a fourth plurality of side scale parameters of the second frame, and to not apply the first stage vector quantizer (141, 143) to the fourth plurality of side scale parameters of the second frame.

[0053] Scale factor decoding At the decoder side, the index is extracted from the bitstream and used to decode and derive the quantized value of the scale factor. A pseudo-code example of the procedure is given below.

[0054] The two-stage decoding procedure is detailed in the following pseudocode. JPEG0007641355000006.jpg156136

[0055] The two-stage decoding procedure is explained in detail in the following pseudocode: JPEG0007641355000007.jpg83111

[0056] The quantized SNS scale factors retrieved from the first stage are refined by decoding the residual in the second stage. The procedure is given in the following pseudocode: JPEG0007641355000008.jpg60137 In an embodiment of the audio inverse quantizer, the quantized audio information items include, for a first frame of the multi-channel audio signal, a first plurality of scale parameters for a first channel of the multi-channel audio signal and a second plurality of scale parameters for a second channel of the multi-channel audio signal. The audio inverse quantizer is configured to apply a first stage vector inverse quantizer (2220) and a second stage vector inverse quantizer (2260) to the first plurality of scale parameters for the first channel of the first frame and the second plurality of scale parameters for the second channel of the first frame. The quantized audio information items include, for a second frame of the multi-channel audio signal, a third plurality of mid-scale parameters and a fourth plurality of side-scale parameters. The audio inverse quantizer is configured to apply a first stage vector inverse quantizer (2220) and a second stage vector inverse quantizer (2260) to a third plurality of mid-scale parameters of the second frame, to apply the second stage vector inverse quantizer (2260) to a fourth plurality of side scale parameters of the second frame, and to not apply the first stage vector inverse quantizer (2220) to the fourth plurality of side scale parameters of the second frame. In an embodiment, the first stage vector inverse quantizer (2220) is configured to receive a first quantization index having a first number of bits for retrieval of the first stage codebook, and the second stage vector inverse quantizer (2260) is configured to receive a second quantization index having a second number of bits for retrieval of the second stage codebook, where the second number of bits is less than or greater than the first number of bits, or, illustratively, the first number of bits is between 4 and 7 bits, and illustratively, the second number of bits is between 3 and 6 bits.

[0057] Regarding the scaling or amplification / weighting of the residual at the encoder side and the scaling or attenuation / weighting at the decoder side, the weighting factor is not calculated separately for each value or split, but a single weight or a few different weights (as an approximation to avoid complexity) is used to scale all parameters. This scaling is, for example, a factor that determines the trade-off between the bitrate savings of coarse quantization (further quantization to zero) and the quantization precision (with its respective spectral distortion), and can be pre-determined at the encoder, so that this pre-determined value does not need to be transmitted to the decoder, but can be fixedly set or initialized at the decoder to save transmission bits. Thus, a higher scaling of the residual requires more bits but has minimal spectral distortion, and when scaling down, additional bits are saved, and if the spectral distortion is kept within an acceptable range, it can serve as a means of additional bitrate savings. In an embodiment, the residual item determiner (142) is configured to divide corresponding differences between the plurality of intermediate quantization items and the audio information item by a predetermined factor smaller than 1 or to multiply corresponding differences between the plurality of intermediate quantization items and the audio information item by a predetermined factor larger than 1, In an embodiment of the audio inverse quantizer, a weighting value or attenuation value is used to multiply the scaling parameter by a weighting value or amplification value, the weighting value being preferably between 0.1 and 0.9, or more preferably between 0.2 and 0.6, or even more preferably between 0.25 and 0.4.

[0058] Advantages of the Preferred Embodiments · Significant bit savings when the two channels are correlated and the SNS parameters are jointly coded. An example of the bits-per-frame savings achieved in the system described in the previous section is given below. o Independent - average 88.1 bits o New Independent - Average 72.0 bits New joints - average 52.1 bits Where: o "Independent" is an MDCT stereo implementation described in [8] that uses SNS [6] for FDNS, where the two channels are coded independently with a two-stage VQ only. 1st stage - 8-bit pre-trained codebook (16 dimensions) 2nd stage - AVQ (variable bitrate) of residual scaled by 4x o "Novel Independent" refers to a previously described embodiment of the invention where the correlation of the two channels is not high enough and they are separately coded using the novel VQ2-stage approach as described above and the residual is scaled by a reduced factor of 2.5. o "New joint" refers to the jointly coded case (also described above), where again in the second stage the residual is scaled by a reduced factor of 2.5. Another advantage of the proposed method is the savings in computational complexity. As shown in [6], the new SNS is more optimal in terms of computational complexity from the LPC-based FDNS described in [5] due to the autocorrelation calculations required to estimate the LPC. Thus, when comparing the computational complexity of the MDCT-based stereo system from [8], where the improved LPC-based FDNS [5] is used, with an implementation where the new SNS [6] replaces the LPC-based method, there is a saving of about 6 WMOPS at a sampling rate of 32 kHz. Furthermore, a new two-stage quantization using VQ for the first stage and AVQ with reduced scale for the second stage achieves some further reduction in computational complexity. In the embodiment described in the previous section, the computational complexity is further reduced by about 1 WMOPS at a sampling rate of 32 kHz, with an acceptable tradeoff in spectral distortion.

[0059] Summary of Preferred Embodiments or Aspects 1. Joint coding of the spectral noise shaping parameters, where a mid / side representation of the parameters is computed and the mid is coded using quantization and entropy coding, and the side is coded using a coarser quantization scheme. 2. Adaptively determine whether the noise shaping parameters should be independently or jointly coded based on channel correlation or coherence. 3. A signaling bit sent to determine whether the parameters are independently or jointly coded. 4. Application example based on MDCT stereo implementation. Signaling using bits with zero side coefficients Social networking sites are used The power spectrum is used to calculate the SNS In the first stage, two splits using 5 bits are used. By adjusting the scaling of the residual in the second stage AVQ, the number of bits for the second stage quantization can be further reduced. Figure 23 shows a comparison of the number of bits for both channels according to the current prior art implementation (described above as "independent"), the new independent implementation according to the second aspect of the invention, and the new joint implementation according to the first aspect of the invention. Figure 23 shows a histogram where the vertical axis represents the frequency of occurrence and the horizontal axis shows bins of the total number of bits for coding the parameters for both channels. Subsequently, a further preferred embodiment is shown in which particular emphasis is given to the calculation of the scale factors for each audio channel and further to the particular application of downsampling and upsampling of the scale parameters, which is applied either before or after the calculation of the joint coding scale parameters as illustrated with respect to Figures 3a, 3b.

[0060] Fig. 11 shows an apparatus for encoding an audio signal 160. The audio signal 160 is preferably available in the time domain, but other representations of the audio signal are primarily useful, such as the predictive domain or any other domain. The apparatus comprises a converter 100, a scale factor calculator 110, a spectral processor 120, a downsampler 130, a scale factor encoder 140 and an output interface 150. The converter 100 is configured for converting the audio signal 160 into a spectral representation. The scale factor calculator 110 is configured for calculating a first set of scale parameters or scale factors from the spectral representation. The other channels are received in block 120 and the scale parameters from the other channels are received by block 140. Throughout this specification, the terms "scale factor" or "scale parameter" are used to refer to the same parameter or value, i.e., a value or parameter used to weight some spectral values ​​after some processing. When this weighting is performed in the linear domain, it is actually a multiplication operation with a scaling factor. However, when the weighting is performed in the logarithmic domain, the weighting operation with the scale factor is actually an addition or subtraction operation. Thus, in the context of this application, scaling does not only mean multiplication or division, but also addition or subtraction depending on the particular domain, or generally each operation in which spectral values ​​are weighted or modified, for example, using a scale factor or scale parameter.

[0061] The downsampler 130 is configured to downsample the first set of scale parameters to obtain a second set of scale parameters, where the second number of scale parameters in the second set of scale parameters is less than the first number of scale parameters in the first set of scale parameters. This is also outlined in the box in FIG. 11 stating that the second number is less than the first number. As shown in FIG. 11 , the scale factor encoder is configured to generate an encoded representation of the second set of scale factors, which encoded representation is forwarded to the output interface 150. Due to the fact that the second set of scale factors has a smaller number of scale factors than the first set of scale factors, the bit rate for transmitting or storing the encoded representation of the second set of scale factors is lower compared to the situation in which the downsampling of the scale factors performed in the downsampler 130 would not have been performed.

[0062] Further, the spectral processor 120 is configured for processing the spectral representation output by the converter 100 in Fig. 11 using a third set of scale parameters, the third set of scale parameters or scale factors having a third number of scale factors greater than the second number of scale factors, where the spectral processor 120 is configured to use the first set of scale factors as already available from the block 110 via line 171 for the purpose of spectral processing. Alternatively, the spectral processor 120 is configured to use the second set of scale factors output by the downsampler 130 for the calculation of the third set of scale factors, as indicated by line 172. In a further implementation, the spectral processor 120 uses the encoded representation output by the scale factor / parameter encoder 140 for the purpose of calculating the third set of scale factors, as indicated by line 173 in Fig. 11. Preferably, the spectral processor 120 does not use the first set of scale factors, but uses the second set of scale factors calculated by the downsampler, or even more preferably uses the coded representation, or in general uses the quantized second set of scale factors and then performs an interpolation operation to interpolate the quantized second set of spectral parameters to obtain a third set of scale parameters having a larger number of scale parameters for the interpolation operation.

[0063] Thus, the coded representation of the second set of scale factors output by block 140 preferably includes either a codebook index or a set of corresponding codebook indexes for the scale parameter codebook used, in other embodiments the coded representation includes a codebook index or a set of codebook indexes, or generally quantized scale parameters for the quantized scale factors obtained when the coded representation is input to a decoder-side vector decoder or any other decoder. Preferably, the spectral processor 120 uses the same set of scale factors that are also available on the decoder side, i.e., uses the quantized second set of scale parameters together with an interpolation operation to finally obtain the third set of scale factors.

[0064] In a preferred embodiment, the third number of scale factors in the third set of scale factors is equal to the first number of scale factors. However, a smaller number of scale factors is also useful. Illustratively, for example, 64 scale factors can be derived in block 110, and then the 64 scale factors can be downsampled to 16 scale factors for transmission. Then, in the spectrum processor 120, an interpolation to 32 scale factors can be performed, not necessarily to 64 scale factors. Alternatively, an interpolation to an even larger number of scale factors, such as more than 64 in some cases, can be performed, as long as the number of scale factors transmitted in the encoded output signal 170 is smaller than the number of scale factors calculated in block 110 or calculated and used in block 120 of FIG. 11. Preferably, the scale factor calculator 110 is configured to perform several operations shown in FIG. 12. These operations refer to the calculation 111 of an amplitude-related measure per band. Here, the spectral representation for one channel is input to block 111. Calculations for the other channels are performed similarly. The preferred amplitude-related measure per band is the energy per band, but other amplitude-related measures may be used as well, for example the sum of the amplitude magnitudes per band, or the sum of the squared amplitudes corresponding to the energy. However, apart from the powers of 2 used to calculate the energy per band, other powers may also be used, such as powers of 3 reflecting the loudness of the signal, and powers other than integers, such as powers of 1.5 or 2.5, may also be used to calculate the amplitude-related measures per band. Powers less than 1.0 may also be used, as long as it is guaranteed that the values ​​processed by such powers are positive values.

[0065] A further operation performed by the scale factor calculator may be inter-band smoothing 112. This inter-band smoothing is preferably used to smooth possible instabilities that may appear in the vector of amplitude-related measures obtained by step 111. Without this smoothing, these instabilities would be amplified when later transformed to the logarithmic domain as shown at 115, especially at spectral values ​​whose energy is close to zero. However, in other embodiments, inter-band smoothing is not performed. A further preferred operation performed by the scale factor calculator 110 is a pre-emphasis operation 113. This pre-emphasis operation has a similar purpose to the pre-emphasis operation used in the LPC-based perceptual filter of the MDCT-based TCX processing, as explained previously with respect to the prior art. This procedure increases the amplitude of the shaped spectrum at low frequencies, which results in a reduction of the quantization noise at low frequencies. However, depending on the implementation, the pre-emphasis operation need not necessarily be performed as any other particular operation.

[0066] A further optional processing operation is the noise floor addition process 114. This procedure improves the quality of signals containing very high spectral dynamics, e.g. glockenspiel, by limiting the amplitude amplification of the shaped spectrum in the valleys, which has the indirect effect of reducing the quantization noise in the peaks at the expense of increasing it in the valleys. Here, the quantization noise is not perceptible in any case due to the masking properties of the human ear, such as absolute hearing threshold, pre-masking, post-masking, or general masking threshold, which indicates that typically very low volume tones that are relatively close in frequency to high volume tones are not perceptible at all, i.e. are completely masked or only roughly perceived by the human hearing mechanism, so this spectral contribution may be very coarsely quantized. However, the noise floor addition operation 114 does not necessarily have to be performed. Further, block 115 indicates a log-like domain transformation. Preferably, the transformation of the output of one of blocks 111, 112, 113, 114 in FIG. 12 is performed in the log-like domain. The log-like domain is a domain in which values ​​close to zero are expanded and high values ​​are compressed. Preferably, the log-like domain is the base 2 domain, but other log-like domains can be used as well. However, the base 2 log-like domain is better for implementation on a fixed-point signal processor. The output of the scale factor calculator 110 is a first set of scale factors.

[0067] As shown in Fig. 12, each of blocks 112 to 115 may be bridged, i.e., for example, the output of block 111 may already be a first set of scale factors. However, all processing operations, in particular logarithmic domain transformation, are preferred. Thus, for example, a scale factor calculator can even be implemented by only performing steps 111 and 115, without the procedures in steps 112 to 114. At the output of block 115, a set of scale parameters for a channel (such as L) is obtained, and sets of scale parameters for other channels (such as R) may also be obtained by similar calculations. The scale factor calculator is therefore configured to perform one or more of the procedures shown in FIG. 12, as indicated by the input / output lines connecting the several blocks.

[0068] FIG. 13 shows again a preferred implementation of the downsampler 130 of FIG. 11 for a single channel. Data for the other channels are calculated in a similar manner. Preferably, low-pass filtering, or generally filtering with a particular window w(k), is performed in step 131, and then a downsampling / decimation operation of the result of the filtering is performed. Due to the fact that the low-pass filtering 131, and in a preferred embodiment the downsampling / decimation operation 132, are both arithmetic operations, the filtering 131 and the downsampling 132 can be performed within a single operation as outlined later. Preferably, the downsampling / decimation operation is performed such that an overlap between the individual groups of scale parameters of the first set of scale parameters is performed. Preferably, an overlap of one scale factor in the filtering operation between the two decimated calculated parameters is performed. Thus, step 131 performs a low-pass filter on the vector of scale parameters before decimation. This low-pass filter has an effect similar to the spreading function used in the psychoacoustic model. It reduces the quantization noise at the peaks at the expense of increasing the quantization noise near the peaks, where it is in any case more perceptually masked at least relative to the quantization noise at the peaks.

[0069] Furthermore, the downsampler further performs a mean value removal 133 and an additional scaling step 134. However, the low-pass filtering operation 131, the mean value removal step 133, and the scaling step 134 are only optional steps. Thus, the downsampler shown in FIG. 13 or shown in FIG. 11 can be implemented to perform only step 132, or to perform two steps shown in FIG. 13, such as step 132 and one of steps 131, 133, and 134. Alternatively, the downsampler can perform all four steps shown in FIG. 13, or only three of the four steps, as long as the downsampling / decimation operation 132 is performed. As outlined in FIG. 13, the audio operations performed by the downsampler in FIG. 13 are performed in the logarithmic domain to obtain better results.

[0070] Fig. 15 shows a preferred implementation of the spectral processor. The spectral processor 120 included in the encoder of Fig. 11 includes an interpolator 121 which receives a quantized second set of scale parameters for each channel, or alternatively for a group of jointly encoded scale parameters, and outputs a third set of scale parameters for the channels for the group of jointly encoded scale parameters, where the third number is greater than the second number and preferably equal to the first number. Furthermore, the spectral processor includes a linear domain transformer 120. Then, in block 123, a spectral shaping is performed using the linear scale parameters on the one hand and the spectral representation obtained by the transformer 100 on the other hand. Preferably, a subsequent temporal noise shaping operation, i.e. a prediction over frequency, is performed to obtain a spectral residual value at the output of block 124, and the TNS side information is transferred to the output interface as indicated by arrow 129.

[0071] Finally, the spectral processor 125, 120b has at least one of a scalar quantizer / encoder, a stereo processing function, and an IGF processing function, etc., configured to receive a single global gain for the entire spectral representation, i.e., the entire frame. Preferably, the global gain is derived in response to certain bitrate considerations. Thus, the global gain is set so that the coded representation of the spectral representation generated by the block 125, 120b meets certain requirements, such as bitrate requirements, quality requirements, or both. The global gain may be calculated iteratively or, possibly, in a feed-forward measure. In general, the global gain is used with a quantizer, where a high global gain typically results in a coarser quantization, and where a low global gain results in a finer quantization. Thus, in other words, when a fixed quantizer is obtained, a high global gain results in a higher quantization step size, and a low global gain results in a smaller quantization step size. However, other quantizers may also be used with the global gain function, e.g., a quantizer with some compression function for higher values, i.e., some non-linear compression function, so that higher values ​​are more compressed than lower values. The above dependency between global gain and quantization coarseness is valid when the global gain is multiplied in the linear domain up to the value before quantization, corresponding to an addition in the logarithmic domain. However, the dependency is reversed if the global gain is applied by division in the linear domain or by subtraction in the logarithmic domain. The same is true when "global gain" represents the inverse value.

[0072] Subsequently, preferred implementations of the individual procedures described with respect to Figs. 11-15 are given. Detailed Step-by-Step Description of the Preferred Embodiment Encoder. Step 1 - Energy per band (111) Energy per band E B (n) is calculated as follows: TIFF0007641355000009.tif18131In the above formula, X(k) is the MDCT coefficient, and N B = 64 is the number of bands and Ind(n) is the band index. The bands are non-uniform and follow the perceptually relevant Bark scale (smaller at low frequencies, larger at high frequencies). Step 2 - Smoothing (112) Energy per band E B (n) is smoothed using: TIFF0007641355000010.tif20126 Note - This step is mainly for Vector E B It is used to smooth out possible instabilities that may appear in (b). If not smoothed, these instabilities will be amplified when transformed into the logarithmic domain, especially in the valleys where the energy is close to zero (see step 5).

[0073] Step 3 - Pre-emphasis (113) Then, the smoothed energy E S (b) is pre-emphasized using: TIFF0007641355000011.tif1177In the above formula, g tilt controls the slope of the pre-emphasis and depends on the sampling frequency: it is, for example, 18 at 16 kHz and 30 at 48 kHz. The pre-emphasis used in this step has the same purpose as the pre-emphasis used in the LPC-based perceptual filter of Prior Art 2, which is to increase the amplitude of the shaped spectrum at low frequencies and result in a reduction of the quantization noise at low frequencies. Step 4 - Noise Floor (114) The noise floor at -40 dB is calculated using E p (b) is added. TIFF0007641355000012.tif796In the above formula, the noise floor is calculated by: TIFF0007641355000013.tif1276 This step improves the quality of signals containing extremely high spectral dynamics, such as glockenspiel, by limiting the amplitude amplification of the shaped spectrum in the valleys, which has the indirect effect of reducing the quantization noise in the peaks at the expense of increasing the quantization noise in the valleys, where the quantization noise is not perceptible anyway.

[0074] Step 5 - Logarithm (115) A transformation to the log domain is then performed using: TIFF0007641355000014.tif1174 Step 6 - Downsampling (131, 132) Next, the vector E L (b) is downsampled by a factor of 4 using: TIFF0007641355000015.tif42111In the above formula, TIFF0007641355000016.tif1353 TIFF0007641355000017.tif7150 Vector E before decimation L Apply a low-pass filter (w(k)) to (b). This low-pass filter has a similar effect to the spreading function used in the psychoacoustic model, i.e. it reduces the quantization noise at the peaks at the expense of increasing the quantization noise near the peaks, where the quantization noise would otherwise be perceptually masked.

[0075] Step 7 - Mean removal and scaling (133, 134) The final scale factor is obtained after mean removal and scaling by 0.85. The TIFF0007641355000018.tif12100 codec has an additional global gain so that the mean can be removed without losing information. Removing the mean also allows for more efficient vector quantization. The 0.85 scaling slightly compresses the amplitude of the noise shaping curve, which has a similar perceptual effect as the spread function mentioned in step 6, i.e. reducing the quantization noise at the peaks and increasing the quantization noise at the valleys. Step 8 - Quantization (141, 142) The scale factors are quantized using vector quantization, which produces an index and a quantized scale factor scfQ(n) that are then packed into the bitstream and sent to the decoder.

[0076] Step 9 - Interpolation (121, 122) The quantized scale factor scfQ(n) is Interpolated using TIFF0007641355000019.tif65138 and converted to the linear domain using: TIFF0007641355000020.tif1075 Interpolation is used to obtain a smooth noise shaping curve and therefore avoid large amplitude jumps between adjacent bands.

[0077] Step 10 - Spectral Shaping (123) SNS scale factor g SNS (b) is the shaped spectrum X s is applied to the MDCT frequency lines separately for each band to generate (k). TIFF0007641355000021.tif11138

[0078] Fig. 18 shows a preferred implementation of an apparatus for decoding an encoded audio signal 250 (stereo signal encoded as L,R or M,S) including information on the encoded spectral representation and on the encoded representation of a second set of scale parameters (separately or jointly encoded). The decoder comprises an input interface 200, a spectral decoder 210 (e.g. performing an IGF processing or an inverse stereo processing or an inverse quantization processing), a scale factor / parameter decoder 220, a spectral processor 230 (e.g. for R,L) and a converter 240 (e.g. for R,L). The input interface 200 is configured to receive the encoded audio signal 250 and to extract the encoded spectral representation which is forwarded to the spectral decoder 210 and to extract the encoded representation of the second set of scale factors which is forwarded to the scale factor decoder 220. Furthermore, the spectral decoder 210 is configured to decode the encoded spectral representation to obtain a decoded spectral representation which is forwarded to the spectral processor 230. The scale factor decoder 220 is configured to decode the encoded second set of scale parameters to obtain a first set of scale parameters which are forwarded to the spectral processor 230. The first set of scale factors has a number of scale factors or scale parameters which is greater than the number of scale factors or scale parameters in the second set. The spectral processor 230 is configured to process the decoded spectral representation using the first set of scale parameters to obtain a scaled spectral representation. The scaled spectral representation is then transformed by the transformer 240 to finally obtain a decoded audio signal 260 which is a stereo signal or a multi-channel signal with more than two channels.

[0079] Preferably, the scale factor decoder 220 is configured to operate in substantially the same manner as described with respect to the spectral processor 120 of FIG. 11 in relation to the calculation of the third set of scale factors or scale parameters, in particular as described with respect to blocks 141 or 142, and in particular as described with respect to blocks 121, 122 of FIG. 15. In particular, the scale factor decoder is configured to perform substantially the same procedure as previously described with respect to step 9 for the interpolation and transformation to the linear domain. Thus, as shown in FIG. 19, the scale factor decoder 220 is configured to apply a decoder codebook 221 to one or more indices per frame representing the encoded scale parameter representation. An interpolation is then performed in block 222, which is substantially the same as the interpolation described with respect to block 121 in FIG. 15. A linear domain transformer 223 is then used, which is substantially the same as the linear domain transformer 122 described with respect to FIG. 15. However, in other implementations, the blocks 221, 222, 223 can operate differently than as described with respect to the corresponding blocks on the encoder side.

[0080] Further, the spectral decoder 210 shown in FIG. 18 or FIG. 19 includes an inverse quantizer / decoder block, which receives the coded spectrum as input and outputs a dequantized spectrum, preferably dequantized using a global gain further transmitted in coded audio signal from the encoder side to the decoder side in coded format. Block 210 may also perform inverse stereo processing, such as IGF processing, or MS decoding. Inverse quantizer / decoder 210 may include, for example, an arithmetic or Huffman decoder function that receives some code as input and outputs quantization indexes representing the spectral values. These quantization indexes are then input to an inverse quantizer together with the global gain, and the output is the dequantized spectral values, which may then undergo TNS processing, such as inverse prediction across frequencies, in TNS decoder processing block 211, although this is optional. In particular, the TNS decoder processing block further receives TNS side information generated by block 124 of FIG. 15, as indicated by line 129. The output of the TNS decoder processing step 211 is input to a spectral shaping block 212 which operates separately for each channel using separate scale factors, where a first set of scale factors calculated by the scale factor decoder is applied to the decoded spectral representation, which may or may not be TNS processed, and the output is a scaled spectral representation for each channel which is then input to the converter 240 of Figure 18.

[0081] Subsequently, further steps of a preferred embodiment of the decoder are described. decoder. · Step 1 - Quantization (221) The vector quantizer index generated in the encoder step 8 is read from the bitstream and used to decode the quantized scale factor scfQ(n). Step 2 - Interpolation (222, 223) Same as encoder step 9. · Step 3 - Spectral Shaping (212) SNS scale factor g SNS (b) The decoded spectrum as outlined by the code This is applied to the quantized MDCT frequency lines for each band separately to produce TIFF0007641355000022.tif810. TIFF0007641355000023.tif9147

[0082] Figures 16 and 17 show a typical encoder / decoder setup, where Figure 16 represents an implementation without TNS processing and Figure 17 shows an implementation including TNS processing. Similar functions shown in Figures 16 and 17 correspond to similar functions in other figures when the same reference numbers are shown. In particular, as shown in Figure 16, an input signal 160, for example a stereo or multi-channel signal, is input to a transform stage 110, after which a spectral processing 120 is performed. In particular, the spectral processing is reflected by an SNS encoder, indicated by reference numbers 123, 110, 130, 140, which indicates that the block SNS encoder implements the functions indicated by these reference numbers. Following the SNS encoder block, a quantization encoding operation 120b, 125 is performed, and the encoded signal is input to a bitstream, as indicated by 180 in Figure 16. The bitstream 180 then occurs at the decoder side, where it is subjected to inverse quantization and decoding as indicated by reference numeral 210, followed by the SNS decoder operations as indicated by blocks 210, 220, 230 in FIG. 18, which finally result in the decoded output signal 260 being obtained, followed by an inverse transform 240.

[0083] FIG. 17 shows a similar representation to that of FIG. 16, but shows that preferably TNS processing is performed following SNS processing at the encoder side, and correspondingly, TNS processing 211 is performed before SNS processing 212 in terms of the processing sequence at the decoder side. Preferably, an additional tool TNS between Spectral Noise Shaping (SNS) and Quantization / Coding (see block diagram below) is used. TNS (Temporal Noise Shaping) also shapes the quantization noise, but also performs time-domain shaping (as opposed to the frequency-domain shaping of SNS). TNS is useful for signals containing sharp attacks and for audio signals. TNS is usually applied between the transform and the SNS (e.g. in AAC), but it is preferable to apply TNS to the shaped spectrum, which avoids some artifacts generated by TNS decoders when the codec operates at low bitrates.

[0084] 20 shows a preferred subdivision into bands of the spectral coefficients or lines obtained by the encoder block 100. In particular, it is shown that the lower bands have a smaller number of spectral lines than the higher bands. In particular, the x-axis in Fig. 20 corresponds to the band index, showing a preferred embodiment of 64 bands, and the y-axis corresponds to the spectral line index, showing 320 spectral coefficients in one frame. In particular, Fig. 20 exemplarily shows the situation in the case of ultra-wideband (SWB) with a sampling frequency of 32 kHz. In the wideband case, the situation for each band is such that one frame results in 160 spectral lines, the sampling frequency is 16 kHz, and therefore in both cases one frame has a time length of 10 milliseconds.

[0085] FIG. 21 shows further details regarding the preferred downsampling performed in downsampler 130 of FIG. 11, or the corresponding upsampling or interpolation performed in scale factor decoder 220 of FIG. 18 or shown in block 222 of FIG. 19. Along the x-axis, an index is given for bands 0 to 63. In particular, there are 64 bands, numbered from 0 to 63. The 16 downsampled points corresponding to scfQ(i) are shown as vertical lines 1100. In particular, Fig. 21 shows how a particular grouping of scale parameters is performed to finally obtain the downsampled points 1100. Illustratively, the first block of four bands consists of (0,1,2,3), and the midpoint of this first block is at 1.5, which is indicated by the item 1100 at index 1.5 along the x-axis. Correspondingly, the second block of four bands is (4,5,6,7) with the midpoint of the second block being 5.5. Windows 1110 correspond to the windows w(k) described above with respect to step 6 downsampling. These windows are centered on the downsampling points, and it can be seen that there is an overlap of one block on each side as previously described.

[0086] The interpolation step 222 in Figure 19 recovers 64 bands from 16 downsampling points. This can be seen in Figure 21 by calculating the position of any of the lines 1120 as a function of the two downsampling points around a particular line 1120, indicated at 1100. The following example illustrates this. The position of the second band is calculated as a function of two perpendicular lines around it (1.5 and 5.5): 2=1.5+1 / 8×(5.5−1.5). Correspondingly, the position of the third band is calculated as a function of the two vertical lines 1100 around it (1.5 and 5.5): 3=1.5+3 / 8×(5.5−1.5). A specific procedure is performed for the first two bands and the last two bands, where an interpolation cannot be performed since either there is no vertical line or the value corresponding to the vertical line 1100 is outside the range from 0 to 63. Therefore, to address this issue, an extrapolation is performed as explained with respect to step 9 - Interpolation outlined above, for the two bands 0, 1 on the one hand and 62 and 63 on the other hand.

[0087] Subsequently, preferred implementations of the converter 100 of FIG. 11 on the one hand and the converter 240 of FIG. 18 on the other hand are described. In particular, Fig. 22a shows a timeline to illustrate the framing performed on the encoder side in the converter 100. Fig. 22b shows a preferred implementation of the converter 100 of Fig. 11 on the encoder side, and Fig. 22c shows a preferred implementation of the converter 240 on the decoder side. The converter 100 on the encoder side is preferably implemented to perform a framing with overlapping frames, such as a 50% overlap, so that frame 2 overlaps with frame 1 and frame 3 overlaps with frames 2 and 4. However, it is preferred to perform a 50% overlap together with the MDCT algorithm, although other overlapping or non-overlapping operations may also be performed. For this purpose, the converter 100 comprises an analysis window 101 and a spectral converter 102 connected afterwards to perform an FFT operation, an MDCT operation or any other kind of time-spectral conversion operation to obtain a sequence of frames corresponding to the sequence of spectral representations as input in FIG. 11 to the block following the converter 100. Correspondingly, the scaled spectral representation is input to a transformer 240 of Fig. 18. In detail, the transformer includes a time transformer 241 performing an inverse FFT operation, an inverse MDCT operation, or a corresponding spectral-to-time conversion operation. The output is inserted into a synthesis window 242, the output of which is input to an overlap-add processor 243 for performing an overlap-add operation to finally obtain a decoded audio signal. In detail, the overlap-add process in block 243 performs a sample-by-sample addition between corresponding samples of, for example, the second half of frame 3 and the first half of frame 4, such that audio sampling values ​​for the overlap between frame 3 and frame 4 as shown by item 1200 in Fig. 22a are obtained. Similar overlap-add operations are performed sample-by-sample to obtain the remaining audio sampling values ​​of the decoded audio output signal.

[0088] It should be noted that, herein, all alternatives or aspects described above, and all aspects defined by the independent claims in the following claims, may be used individually, i.e., without alternatives or purposes other than those contemplated in the alternatives, purposes or independent claims. However, in other embodiments, two or more of the alternatives or aspects or independent claims may be combined with each other, and in other embodiments, all aspects or alternatives and all independent claims may be combined with each other. Although further aspects have been described above, the appended claims show two different aspects, namely an audio decoder, an audio encoder and related methods using joint coding of scale parameters for channels of a multi-channel audio signal, or an audio quantizer, an audio inverse quantizer or related methods. These two aspects may possibly be combined or used separately, and the invention according to these aspects is applicable to other applications of audio processing different from the specific application described above.

[0089] Reference is further made to additional Figures 3a, 3b, 4a, 4b, 5, 6, 8a, 8b showing the first embodiment, and Figures 9a, 9b showing the second embodiment, as well as Figures 7a, 7b showing the second embodiment applied within the first embodiment. The encoded signals of the present invention may be stored on a digital or non-transitory storage medium or may be transmitted over a transmission medium, such as a wireless transmission medium or a wired transmission medium such as the Internet. Although some aspects have been described in the context of an apparatus, it will be apparent that these aspects also represent descriptions of corresponding methods, where a block or device corresponds to a method step or feature of a method step, and similarly, aspects described in the context of a method step also represent descriptions of a corresponding block or item or feature of a corresponding apparatus. Depending on specific implementation requirements, embodiments of the present invention may be implemented in hardware or software, such as using digital storage media, e.g. floppy disks, DVDs, CDs, ROMs, PROMs, EPROMs, EEPROMs or flash memories, that store electronically readable control signals, which cooperate (or can cooperate) with a programmable computer system to execute the respective methods.

[0090] Some embodiments according to the invention include a data carrier having electronically readable control signals capable of cooperating with a programmable computer system to cause one of the methods described herein to be performed. Generally, embodiments of the present invention may be implemented as a computer program product having program code operable to perform one of the methods when the computer program product runs on a computer. The program code may for example be stored on a machine readable carrier. Other embodiments comprise the computer program for performing one of the methods described herein, stored on a machine readable carrier or a non-transitory storage medium. In other words, an embodiment of the inventive method is, therefore, a computer program having 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 therefore a data carrier (or a digital storage medium or a computer readable medium) having recorded thereon a computer program for performing one of the methods described herein.

[0091] A further embodiment of the inventive method is therefore a data stream or a sequence of signals representing the computer program for performing one of the methods described herein. The data stream or the sequence of signals can for example be adapted to be transferred via a data communication connection, for example via the Internet. A further embodiment comprises a processing means, for example a computer, or a programmable logic device, configured to or adapted to perform one of the methods described herein. A further embodiment comprises a computer having installed thereon the computer program for performing one of the methods described herein. In some embodiments, a programmable logic device (e.g., a field programmable gate array) may be used to perform some or all of the functions of the methods described herein. In some embodiments, a field programmable gate array may cooperate with a microprocessor to perform one of the methods described herein. In general, the methods are preferably performed by any hardware apparatus. The above-described embodiments are merely illustrative of the principles of the present invention. It is to be understood that modifications and variations of the arrangements and details described herein are apparent to those skilled in the art. It is therefore intended to be limited only by the scope of the claims that follow, and not by the specific details presented as descriptions and illustrations of the embodiments herein.

[0092] Subsequently, further embodiments / examples are summarized. 1. An audio decoder for decoding an encoded audio signal including multi-channel audio data including data for two or more audio channels and information regarding a joint encoding scale parameter, comprising: a scale parameter decoder (220) for decoding the information about the jointly encoded scale parameters to obtain a first set of scale parameters for a first channel of a decoded audio signal and a second set of scale parameters for a second channel of the decoded audio signal; a signal processor (210, 212, 230) for applying the first set of scale parameters to a first channel representation derived from the multi-channel audio data and for applying the second set of scale parameters to a second channel representation derived from the multi-channel audio data to obtain the first channel and the second channel of the decoded audio signal; the joint coding scale parameters include information about a first group of joint coding scale parameters and information about a second group of joint coding scale parameters; the scale parameter decoder (220) is configured to combine jointly coded scale parameters of the first group and jointly coded scale parameters of the second group using a first combining rule to obtain scale parameters of the first set of scale parameters, and using a second combining rule different from the first combining rule to obtain scale parameters of the second set of scale parameters.

[0093] 2. An audio decoder as described in Example 1, wherein the first group of jointly coded scale parameters includes mid-scale parameters and the second group of jointly coded scale parameters includes side-scale parameters, and the scale parameter decoder (220) is configured to use addition in the first combination rule and to use subtraction in the second combination rule. 3. The encoded audio signal is organized into a sequence of frames, a first frame containing the multi-channel audio data and the information regarding the joint encoded scale parameter, and a second frame containing separate encoded scale parameter information; 3. The audio decoder of claim 1 or 2, wherein the scale parameter decoder (220) is configured to detect that the second frame includes the separately encoded scale parameter information and to calculate the first set of scale parameters and the second set of scale parameters. 4. The first frame and the second frame each include state side information indicating that in a first state, the first frame includes the information regarding the joint coding scale parameter, and in a second state, the second frame includes the separate coding scale parameter information; 4. An audio decoder as described in Example 3, wherein the scale parameter decoder (220) is configured to read the state side information of the second frame and detect that the second frame includes the separately encoded scale parameter information based on the read state side information, or to read the state side information of the first frame and detect that the first frame includes the information regarding the jointly encoded scale parameter using the read state side information.

[0094] 5. The signal processor (210, 212, 230) is configured to decode the multi-channel audio data to derive the first channel representation and the second channel representation, the first channel representation and the second channel representation being spectral domain representations having spectral sampling values; the signal processor (210, 212, 230) is configured to apply each scale parameter of the first set and the second set to a corresponding plurality of the spectral sampling values ​​to obtain a shaped spectral representation of the first channel and a shaped spectral representation of the second channel. 5. An audio decoder according to any one of embodiments 1 to 4. 6. An audio decoder as described in Example 5, wherein the signal processor (210, 212, 230) is configured to convert the shaped spectral representation of the first channel and the shaped spectral representation of the second channel to the time domain to obtain a time domain representation of the first channel and a time domain representation of the second channel of the decoded audio signal.

[0095] 7. The first channel representation includes a first number of bands, and the first set of scale parameters includes a second number of scale parameters, the second number being less than the first number; the signal processor (210, 212, 230) is configured to interpolate the second number of scale parameters to obtain a number of interpolated scale parameters that is equal to or greater than the first number of bands, and the signal processor (210, 212, 230) is configured to scale the first channel representation using the interpolated scale parameters; or or the first channel representation comprises a first number of bands, and the information about the first group of joint coding scale parameters comprises a second number of joint coding scale parameters, the second number being less than the first number; the scale parameter decoder (220) is configured to interpolate the second number of jointly coded scale parameters to obtain a number of interpolated jointly coded scale parameters that is equal to or greater than the first number of bands; 7. An audio decoder according to any one of claims 1 to 6, wherein the scale parameter decoder (220) is configured to process the interpolated jointly encoded scale parameters to determine the first set of scale parameters and the second set of scale parameters.

[0096] 8. The encoded audio signal is organized into a sequence of frames, and the information regarding the second group of jointly coded scale parameters includes zero-side information at a particular frame, and the scale parameter decoder (220) is configured to detect the zero-side information to determine that the second group of jointly coded scale parameters is all zero for the particular frame; 8. An audio decoder according to any one of claims 1 to 7, wherein the scale parameter decoder (220) is configured to derive the scale parameters of the first set of scale parameters and the second set of scale parameters only from the first group of jointly coded scale parameters or to set the scale parameters of the combination of the jointly coded scale parameters of the first group and the jointly coded scale parameters of the second group to zero values ​​or values ​​less than a noise threshold. 9. The scale parameter decoder (220) dequantizing the information relating to the first group of jointly coded scale parameters using a first dequantization mode; dequantizing the information related to the second group of jointly coded scale parameters using a second dequantization mode, the second dequantization mode being different from the first dequantization mode; configured to: 9. An audio decoder according to any one of Examples 1 to 8.

[0097] 10. An audio decoder as described in Example 9, wherein the scale parameter decoder (220) is configured to use the second inverse quantization mode associated with a lower or higher quantization precision than the first inverse quantization mode. 11. The scale parameter decoder (220) uses a first inverse quantization stage (2220) and a second inverse quantization stage (2260) and a combiner (2240) as the first inverse quantization mode, the combiner (2240) receiving as input the result of the first inverse quantization stage (2220) and the result of the second inverse quantization stage (2260); 11. An audio decoder as described in Example 9 or 10, configured to use the second inverse quantization stage (2220) of the first inverse quantization mode as the second inverse quantization mode, the second inverse quantization stage receiving as input the information regarding the second group of joint coding scale parameters. 12. An audio decoder as described in Example 11, wherein the first inverse quantization stage (2220) is a vector inverse quantization stage and the second inverse quantization stage (2260) is an algebraic vector inverse quantization stage, or the first inverse quantization stage (2220) is a fixed rate inverse quantization stage and the second inverse quantization stage (2260) is a variable rate inverse quantization stage.

[0098] 13. The information on the first group of jointly coded scale parameters includes, for a frame of the coded audio signal, two or more indices, and the information on the second group of jointly coded scale parameters includes, for a frame of the coded audio signal, a single index or a smaller or equal number of indices than in the first group; 13. An audio decoder according to embodiment 11 or 12, wherein the scale parameter decoder (220) is configured to determine, in the first inverse quantization stage (2220), for example for each index of the two or more indexes, an intermediate joint coded scale parameter of the first group; and the scale parameter decoder (220) is configured to calculate, in the second inverse quantization stage (2260), for example from the single or a smaller or equal number of indexes of the information relating to the first group of joint coded scale parameters, the first group of joint coded scale parameters from the intermediate joint coded scale parameter of the first group and the residual joint coded scale parameter of the first group by the combiner (2240).

[0099] 14. An audio decoder described in any one of Examples 11 to 13, wherein the first inverse quantization stage (2220) includes using an index for a first codebook having a first number of entries or using an index representing a first precision, and the second inverse quantization stage (2260) includes using an index for a second codebook having a second number of entries or using an index representing a second precision, the second number being less or more than the first number, or the second precision being lower or higher than the first precision. 15. The information about the second group of jointly coded scale parameters indicates that the second group of jointly coded scale parameters are all zero or a specific value for a frame of the encoded audio signal, and the scale parameter decoder (220) is configured to use jointly coded scale parameters that are zero or the specific value or a combined jointly coded scale parameter in the combination using the first rule or the second rule; or 15. An audio decoder according to any one of claims 1 to 14, wherein for the frames containing information of the all-zero or specific values, the scale parameter decoder (220) is configured to determine the second set of scale parameters using only the first group of jointly encoded scale parameters without using a combination operation.

[0100] 16. An audio decoder as described in Example 9 or 10, wherein the scale parameter decoder (220) is configured to use the first inverse quantization stage (2220), the second inverse quantization stage (2260), and the combiner (2240) as the first inverse quantization mode, wherein the combiner (2240) receives and uses the results of the first inverse quantization stage (2220) and the second inverse quantization stage (2260) as inputs, and to use the first inverse quantization stage (2220) of the first inverse quantization mode as the second inverse quantization stage.

[0101] 17. An audio encoder for encoding a multi-channel audio signal containing two or more channels, comprising: a scale parameter calculator (140) for calculating a first group of jointly encoded scale parameters and a second group of jointly encoded scale parameters from a first set of scale parameters for a first channel of the multi-channel audio signal and a second set of scale parameters for a second channel of the multi-channel audio signal; a signal processor (120) for applying the first set of scale parameters to the first channel of the multi-channel audio signal and for applying the second set of scale parameters to the second channel of the multi-channel audio signal to derive multi-channel audio data; an encoded signal former (1480, 1500) for using the multi-channel audio data, information about the first group of jointly encoded scale parameters, and information about the second group of jointly encoded scale parameters to obtain an encoded multi-channel audio signal; an audio encoder.

[0102] 18. The signal processor (120) in the application: encoding the first group of jointly coded scale parameters and the second group of jointly coded scale parameters to obtain the information on the first group of jointly coded scale parameters and the information on the second group of jointly coded scale parameters; locally decoding the information about the first and second groups of jointly encoded scale parameters to obtain a locally decoded first set of scale parameters and a locally decoded second set of scale parameters; scaling the first channel using the locally decoded first set of scale parameters and scaling the second channel using the locally decoded second set of scale parameters. or The signal processor (120), in the application, quantizing the first group of joint coded scale parameters and the second group of joint coded scale parameters to obtain a quantized first group of joint coded scale parameters and a quantized second group of joint coded scale parameters; locally decoding the quantized first and second groups of jointly encoded scale parameters to obtain a locally decoded first set of scale parameters and a locally decoded second set of scale parameters; scaling the first channel using the locally decoded first set of scale parameters and scaling the second channel using the locally decoded second set of scale parameters. 18. An audio encoder as described in Example 17, configured to:

[0103] 19. The scale parameter calculator (140) is configured to combine scale parameters of the first set of scale parameters and scale parameters of the second set of scale parameters using a first combining rule to obtain joint coded scale parameters of the first group of joint coded scale parameters, and using a second combining rule different from the first combining rule to obtain joint coded scale parameters of the second group of joint coded scale parameters. 19. An audio encoder as recited in Example 17 or 18. 20. An audio encoder as described in Example 19, wherein the first group of jointly encoded scale parameters includes mid-scale parameters, the second group of jointly encoded scale parameters includes side-scale parameters, and the scale parameter calculator (140) is configured to use addition in the first combination rule and to use subtraction in the second combination rule.

[0104] 21. The scale parameter calculator is configured to process a sequence of frames of the multi-channel audio signal; The scale parameter calculator (140): calculating first and second groups of jointly encoded scale parameters for a first frame of the sequence of frames; analyzing the second frame of the sequence of frames to determine a separate coding mode for the second frame; An audio encoder according to any one of Examples 17 to 20, wherein the encoded signal former (1480, 1500) is configured to introduce into the encoded audio signal state side information indicating a separate encoding mode for the second frame or a joint encoding mode for the first frame, and information regarding the first set and the second set of separate encoding scale parameters for the second frame.

[0105] 22. The scale parameter calculator (140): calculating the first set of scale parameters for the first channel and the second set of scale parameters for the second channel; downsampling the first and second sets of scale parameters to obtain a downsampled first set and a downsampled second set; combining scale parameters from the downsampled first set and the downsampled second set using different combining rules to obtain joint coded scale parameters of the first group and joint coded scale parameters of the second group; or The scale parameter calculator (140): calculating the first set of sales parameters for the first channel and the second set of scale parameters for the second channel; combining scale parameters from the first set and scale parameters from the second set using different combining rules to obtain jointly encoded scale parameters of the first group and jointly encoded scale parameters of the second group; downsampling the first group of joint coded scale parameters to obtain a downsampled first group of joint coded scale parameters, and downsampling the second group of joint coded scale parameters to obtain a downsampled second group of joint coded scale parameters, An audio encoder according to any one of Examples 17 to 21, wherein the downsampled first group and the downsampled second group represent the information regarding the first group of joint encoding scale parameters and the information regarding the second group of joint encoding scale parameters.

[0106] 23. The scale parameter calculator (140) is configured to calculate a similarity between the first channel and the second channel in the second frame, and determine the separate encoding mode if the calculated similarity has a first relationship with a threshold, or determine the joint encoding mode if the calculated similarity has a different second relationship with the threshold. 23. An audio encoder as recited in Example 21 or 22. 24. The scale parameter calculator (140) calculating, for each band, a difference between the scale parameters of the first set and the scale parameters of the second set for the second frame; processing each difference of the second frame to eliminate a negative sign to obtain a processed difference of the second frame; combining the processed differences to obtain a similarity measure; and comparing said similarity to a threshold; When the similarity is greater than the threshold, the separate coding mode is determined as a priority, or when the similarity is less than the threshold, the joint coding mode is determined as a priority. 24. An audio encoder as described in Example 23, configured to:

[0107] 25. The signal processor (120) obtaining one or more first quantization indexes as a first-stage result, and quantizing the first group of joint coding scale parameters using a first-stage quantization function (141, 143) to obtain an intermediate first group of joint coding scale parameters; calculating (142) a first group of jointly coded scaled parameters residuals from the first group of jointly coded scaled parameters and the intermediate first group of jointly coded scaled parameters; quantizing the first group of residuals of the jointly coded scale parameters using a second-stage quantization function (145) to obtain one or more quantization indexes as a second-stage result; An audio encoder according to any one of Examples 17 to 24, configured to perform the following:

[0108] 26. The signal processor (120) is configured to quantize the second group of jointly coded scale parameters using a single-stage quantization function to obtain one or more quantization indexes as a single-stage result; or the signal processor (120) is configured for quantizing the first group of jointly coded scale parameters using at least a first stage quantization function and a second stage quantization function, and the signal processor (120) is configured for quantizing the second group of jointly coded scale parameters using a single stage quantization function, the single stage quantization function being selected from the first stage quantization function and the second stage quantization function. 26. An audio encoder according to any one of Examples 17 to 25.

[0109] 27. The scale parameter calculator (140): obtaining one or more first quantization indexes as a first stage result, and quantizing the first set of scale parameters using a first stage quantization function (141, 143) to obtain an intermediate first set of scale parameters; calculating (142) a first set of scale parameter residuals from the first set of scale parameters and the intermediate first set of scale parameters; quantizing said first set of residuals of scale parameters using a second-stage quantization function (145) to obtain one or more quantization indexes as a second-stage result; or The scale parameter calculator (140): obtaining one or more first quantization indexes as a first stage result, and quantizing said second set of scale parameters using a first stage quantization function (141, 143) to obtain an intermediate second set of scale parameters; calculating (142) a second set of scale parameter residuals from the second set of scale parameters and the intermediate second set of scale parameters; quantizing the second set of residuals of scale parameters using a second-stage quantization function (145) to obtain one or more quantization indexes as a second-stage result; 27. An audio encoder according to any one of Examples 21 to 26, configured to perform the following:

[0110] 28. The second stage quantization function (145) uses an amplification or weighting value less than 1 to increase the first group of residuals of joint coding scaling parameters or the first or second set of residuals of scale parameters before performing vector quantization, and the vector quantization is performed using the increased residual values; and / or Exemplarily, the weighting or amplification value is used to divide the scaling parameter by the weighting or amplification value, the weighting value being preferably between 0.1 and 0.9, or more preferably between 0.2 and 0.6, or even more preferably between 0.25 and 0.4; and / or the same amplification value is used for all scaling parameters of the first group of residuals of jointly encoded scaling parameters or the first set of residuals of scale parameters or the second set of residuals of scale parameters. 28. An audio encoder as recited in Example 25 or 27.

[0111] 29. The first stage quantization function (141, 143) includes at least one codebook having a first number of entries corresponding to a first size of the one or more quantization indexes; the second stage quantization function (145) or the single stage quantization function includes at least one codebook having a second number of entries corresponding to a second size of the one or more quantization indexes; the first number is greater than or less than the second number, or the first size is greater than or less than the second size, or the first stage quantization function (141, 143) is a fixed rate quantization function, and the second stage quantization function (145) is a variable rate quantization function; 29. An audio encoder according to any one of Examples 25 to 28.

[0112] 30. The scale parameter calculator (140): receiving a first MDCT representation for the first channel and a second MDCT representation for the second channel; receiving a first MDST representation for the first channel and a second MDST representation for the second channel; calculating a first power spectrum for the first channel from the first MDCT representation and the first MDST representation, and calculating a second power spectrum for the second channel from the second MDCT representation and the second MDST representation; calculating the first set of scale parameters for the first channel from the first power spectrum and the second set of scale parameters for the second channel from the second power spectrum; 30. An audio encoder as described in any one of Examples 15 to 29, configured to perform the following.

[0113] 31. The signal processor (120) is configured to scale the first MDCT representation using information derived from the first set of scale parameters and to scale the second MDCT representation using information derived from the second set of scale parameters. 31. An audio encoder as described in Example 30. 32. The signal processor (120) is configured to further process the scaled first channel representation and the scaled second channel representation using joint multi-channel processing to derive a multi-channel processed representation of the multi-channel audio signal, optionally further processing using a spectral band replication process or an intelligent gap filling process or a bandwidth extension process, and quantizing and encoding the representations of the channels of the multi-channel audio signal to obtain the multi-channel audio data. 32. An audio encoder according to any one of Examples 17 to 31.

[0114] 33. An audio encoder according to any one of embodiments 17 to 34, configured to determine, for a frame of the multi-channel audio signal, the information regarding the second group of joint coding scale parameters as all-zero or all-specific value information indicating the same value or zero value for all joint coding scale parameters of the frame, and configured for the coded signal former (1480, 1500) to use the all-zero or all-specific value information to obtain the coded multi-channel audio signal. 34. The scale parameter calculator (140): calculating the first group of jointly encoded scale parameters and the second group of jointly encoded scale parameters for a first frame; and calculating the first group of jointly encoded scale parameters for a second frame; In the second frame, the joint coding scale parameter is not calculated or coded; An audio encoder according to any one of embodiments 17 to 34, wherein the encoded signal former (1480, 1500) is configured to use a flag as the information regarding the second group of joint encoding scale parameters, the flag indicating that in the second frame, none of the joint encoding scale parameters of the second group are included in the encoded multi-channel audio signal.

[0115] 35. A method for decoding an encoded audio signal containing multi-channel audio data including data for two or more audio channels and information regarding a joint encoding scale parameter, comprising: decoding the information about the jointly encoded scale parameters to obtain a first set of scale parameters for a first channel of a decoded audio signal and a second set of scale parameters for a second channel of the decoded audio signal; applying the first set of scale parameters to a first channel representation derived from the multi-channel audio data and applying the second set of scale parameters to a second channel representation derived from the multi-channel audio data to obtain the first channel and the second channel of the decoded audio signal; the joint coding scale parameters include information about a first group of joint coding scale parameters and information about a second group of joint coding scale parameters; the decoding includes combining jointly coded scale parameters of the first group and jointly coded scale parameters of the second group using a first combining rule to obtain scale parameters of the first set of scale parameters and using a second combining rule different from the first combining rule to obtain scale parameters of the second set of scale parameters.

[0116] 36. A method for encoding a multi-channel audio signal containing two or more channels, comprising the steps of: calculating a first group of jointly encoded scale parameters and a second group of jointly encoded scale parameters from a first set of scale parameters for a first channel of the multi-channel audio signal and a second set of scale parameters for a second channel of the multi-channel audio signal; applying the first set of scale parameters to the first channel of the multi-channel audio signal and applying the second set of scale parameters to the second channel of the multi-channel audio signal to derive multi-channel audio data; using the multi-channel audio data, the information about the first group of joint coding scale parameters, and the information about the second group of joint coding scale parameters to obtain an encoded multi-channel audio signal. A method comprising: 37. A computer program for carrying out the method according to example 31 or example 32 when running on a computer or processor. Audio decoders, audio encoders, and related methods using joint coding of scale parameters for channels of a multi-channel audio signal.

[0117] References [1]ISO / IEC11172-3, Information technology-Coding of moving pictures and associated audio for digital storage media at up to about 1,5Mbit / s-Part3:Audio, 1993 [2]ISO / IEC13818-7, Information technology-Generic coding of moving pictures and associated audio information-Part7:Advanced Audio Coding(AAC), 2003 [3]ISO / IEC23003-3, Information technology-MPEG audio technologies-Part3: Unified speech and audio coding [4]3GPP TS26.445, Codec for Enhanced Voice Services(EVS);Detailed algorithmic description [5] G. Markovic, G. Fuchs, N. Rettelbach, C. Helmrich and B. Schubert, "LINEAR PREDICTION BASED CODING SCHEME USING SPECTRAL DOMAIN NOISE SHAPNG," U.S. Patent No. 9,595,262(B2), March 14, 2017. [6] E. Ravelli, M. Schnell, C. Benndorf, M. Lutzky and M. Dietz, “Apparatus and method for encoding and decoding an audio signal using downsampling or interpolation of scale parameters”, International Publication No. 2019091904(A1), November 5, 2018. [7] A. Biswas, Advances in Perceptual Stereo Audio Coding Using Linear Prediction Techniques, Eindhoven: Technical University of Eindhoven, 2017 [8] G. Markovic, E. Ravelli, M. Schnell, S. Dohla, W. Jaegars, M. Dietz, C. Heimrich, E. Fotopoulou, M. Multrus, S. Bayer, G. Fuchs and J. Herre, “APPARATUS AND METHOD FOR MDCT M / S STEREO WITH GLOBAL ILD WITH IMPROVED MID / SIDE DECISION” International Publication No. 2017EP5117

Claims

1. 1. An audio quantizer for quantizing a plurality of audio information items, comprising: a first stage vector quantizer (141, 143) for quantizing a plurality of audio information items to determine a first stage vector quantization result and a plurality of intermediate quantization items corresponding to the first stage vector quantization result; a residual item determiner (142) for calculating a plurality of residual items from said plurality of intermediate quantization items and said plurality of audio information items; a second stage vector quantizer (145) for quantizing the plurality of residual items to obtain a second stage vector quantization result, the first stage vector quantization result and the second stage vector quantization result being a quantized representation of the plurality of audio information items; Including, said audio information item comprising, for a first frame of a multi-channel audio signal, a first plurality of scale parameters for a first channel of the multi-channel audio signal and a second plurality of scale parameters for a second channel of the multi-channel audio signal; the audio quantizer is configured to apply a first stage vector quantizer (141, 143) and a second stage vector quantizer (145) to the first plurality of scale parameters for the first channel of the first frame and the second plurality of scale parameters for the second channel of the first frame; said audio information item comprising, for a second frame of said multi-channel audio signal, a third plurality of mid-scale parameters and a fourth plurality of side-scale parameters; the audio quantizer is configured to apply the first stage vector quantizer (141, 143) and the second stage vector quantizer (145) to the third plurality of mid-scale parameters of the second frame, apply the second stage vector quantizer (145) to the fourth plurality of side scale parameters of the second frame, and not apply the first stage vector quantizer (141, 143) to the fourth plurality of side scale parameters of the second frame. Audio quantizer.

2. 2. Audio quantizer according to claim 1, wherein the residual item determiner (142) is arranged to calculate, for each residual item, a difference between a corresponding audio information item and a corresponding intermediate quantization item.

3. 3. The audio quantizer of claim 1, wherein the residual item determiner (142) is configured to, for each residual item, amplify or weight a difference between a corresponding audio information item and a corresponding intermediate quantization item such that the residual items are larger than the corresponding differences, or to amplify or weight the audio information items and / or the intermediate quantization items before calculating the differences between the amplified items to obtain the residual item.

4. the residual item determiner (142) is configured to divide corresponding differences between the plurality of intermediate quantization items and the audio information item by a predetermined factor smaller than 1 or to multiply corresponding differences between the plurality of intermediate quantization items and the audio information item by a predetermined factor larger than 1. An audio quantizer according to any one of claims 1 to 3.

5. the first stage vector quantizer (141, 143) is configured to perform quantization with a first quantization precision and the second stage vector quantizer (145) is configured to perform quantization with a second quantization precision, the second quantization precision being lower or higher than the first quantization precision; or the first stage vector quantizer (141, 143) is configured to perform fixed rate quantization and the second stage vector quantizer (145) is configured to perform variable rate quantization; An audio quantizer according to any one of claims 1 to 4.

6. 6. The audio quantizer of claim 1, wherein the first stage vector quantizer (141, 143) is configured to use a first stage codebook having a first number of entries and the second stage vector quantizer (145) is configured to use a second stage codebook having a second number of entries, the second number of entries being less than or greater than the first number of entries.

7. the audio information item being scale parameters for a frame of the multi-channel audio signal usable in the time domain for scaling time-domain audio samples of the multi-channel audio signal or usable in the spectral domain for scaling spectral-domain audio samples of the multi-channel audio signal, each scale parameter being usable for scaling at least two time-domain or spectral-domain audio samples, the frame comprising a first number of scale parameters; the first stage vector quantizer (141, 143) is configured to perform a split of the first number of scale parameters into two or more sets of scale parameters including a first set of scale parameters and a second set of scale parameters, and the first stage vector quantizer (141, 143) is configured to determine a quantization index for each set of scale parameters to obtain a plurality of quantization indexes representing the first stage vector quantization result. An audio quantizer according to any one of claims 1 to 6.

8. 8. The audio quantizer of claim 7, wherein the first stage vector quantizer (141, 143) is configured to combine a first quantization index for the first set of scale parameters and a second quantization index for the second set of scale parameters to obtain a single index as the first stage vector quantization result.

9. the first stage vector quantizer (141, 143) is configured to multiply one of the first quantization index and the second quantization index by a number corresponding to a number of bits of the first quantization index and the second quantization index to obtain the single index, and to add the multiplied index and the unmultiplied index.

9. An audio quantizer as claimed in claim 8.

10. the second stage vector quantizer (145) is an algebraic vector quantizer, and each quantization index includes a base codebook index and a Voronoi extension index; An audio quantizer according to any one of the preceding claims.

11. said first stage vector quantizer (141, 143) is adapted to perform a first split of said plurality of audio information items; the second stage vector quantizer (145) is configured to perform a second split of the plurality of residual terms; said first split resulting in a first number of subsets of said audio information items and said second split resulting in a second number of subsets of residual items, said first number of subsets being equal to said second number of subsets. An audio quantizer according to any one of claims 1 to 10.

12. the first stage vector quantizer (141, 143) is configured to output a first quantization index having a first number of bits from a first codebook search; the second stage vector quantizer (145) is configured to output a second quantization index having a second number of bits from a second codebook search, the second number of bits being less than or greater than the first number of bits; An audio quantizer according to any one of the preceding claims.

13. the first number of bits is between 4 and 7 bits, and the second number of bits is between 3 and 6 bits.

13. An audio quantizer as claimed in claim 12.

14. the residual item determiner (142) is configured to amplify or weight the fourth plurality of side scale parameters for the second frame, and the second stage vector quantizer (145) is configured to process the amplified or weighted side scale parameters for the second frame of the multi-channel audio signal. The audio quantizer of claim 1 .

15. 1. An audio inverse quantizer for inverse quantizing a plurality of quantized audio information items, comprising: a first stage vector dequantizer (2220) for dequantizing first stage vector quantization results contained in said quantized audio information items to obtain a plurality of intermediate quantized audio information items; a second stage vector dequantizer (2260) for dequantizing second stage vector quantization results contained in said quantized audio information items to obtain a plurality of residual items; a combiner (2240) for combining said intermediate quantized audio information items and said residual items to obtain a plurality of dequantized audio information items; Including, the quantized audio information items comprise, for a first frame of a multi-channel audio signal, a first plurality of scale parameters for a first channel of the multi-channel audio signal and a second plurality of scale parameters for a second channel of the multi-channel audio signal; the audio inverse quantizer is configured to apply the first stage vector inverse quantizer (2220) and the second stage vector inverse quantizer (2260) to the first plurality of scale parameters for the first channel of the first frame and the second plurality of scale parameters for the second channel of the first frame; the quantized audio information items include a third plurality of mid-scale parameters and a fourth plurality of side-scale parameters for a second frame of the multi-channel audio signal; the audio inverse quantizer is configured to apply the first stage vector inverse quantizer (2220) and the second stage vector inverse quantizer (2260) to the third plurality of mid-scale parameters of the second frame, apply the second stage vector inverse quantizer (2260) to the fourth plurality of side scale parameters of the second frame, and not apply the first stage vector inverse quantizer (2220) to the fourth plurality of side scale parameters of the second frame. Audio inverse quantizer.

16. 16. Audio inverse quantizer according to claim 15, wherein the combiner (2240) is configured to calculate, for each inverse quantized information item, the sum of a corresponding intermediate quantized audio information item and a corresponding residual item.

17. the combiner (2240) is configured to attenuate or weight a plurality of residual terms such that a attenuated residual term is smaller than a corresponding residual term before performing the attenuation; the combiner (2240) is adapted to add the attenuated residual item to the corresponding intermediate quantized audio information item, or or the combiner (2240) is configured to use a damping or weighting value smaller than 1 to damp the residual terms or joint coding scaling parameters before performing the combining, the combining being performed using damped residual values; and / or Exemplarily, the weighting value or the attenuation value is used to multiply the scaling parameter by a weighting value or an amplification value, the weighting value being preferably between 0.1 and 0.9, or more preferably between 0.2 and 0.6, or even more preferably between 0.25 and 0.4; and / or the same attenuation value or said weighting value is used for all scaling parameters or any joint coding scaling parameters of the plurality of residual items, 17. An audio inverse quantizer according to claim 15 or 16.

18. 20. The audio inverse quantizer of claim 17, wherein the combiner (2240) is configured to multiply corresponding residual terms by a weighting factor smaller than one or to divide corresponding residual terms by a weighting factor larger than one.

19. the first stage vector inverse quantizer (2220) is configured to perform inverse quantization with a first precision; the second stage vector dequantizer (2260) is configured to perform dequantization with a second precision, the second precision being lower or higher than the first precision; An audio inverse quantizer according to any one of claims 15 to 18.

20. the first stage vector dequantizer (2220) is configured to use a first stage codebook having a first number of entries, and the second stage vector dequantizer (2260) is configured to use a second stage codebook having a second number of entries, the second number of entries being less than or greater than the first number of entries, or the first-stage vector inverse quantizer (2220) is configured to receive a first quantization index having a first number of bits for retrieval of the first-stage codebook; the second stage vector inverse quantizer (2260) is configured to receive a second quantization index having a second number of bits for retrieving the second stage codebook, the second number of bits being less than or greater than the first number of bits, or, exemplarily, the first number of bits is between 4 and 7 bits, and exemplarily, the second number of bits is between 3 and 6 bits; An audio inverse quantizer according to any one of claims 15 to 19.

21. the dequantized audio information items being scale parameters for a frame of the multi-channel audio signal usable in the time domain for scaling time-domain audio samples of the multi-channel audio signal or usable in the spectral domain for scaling spectral-domain audio samples of the multi-channel audio signal, each scale parameter being usable for scaling at least two time-domain or spectral-domain audio samples, the frame comprising a first number of scale parameters; the first-stage vector dequantizer (2220) is configured to determine a first set of scale parameters and a second set of scale parameters from two or more result indexes for the first-stage vector quantization result; the first stage vector inverse quantizer (2220) or the combiner (2240) is configured to combine the first set of scale parameters and the second set of scale parameters into a vector to obtain the first number of intermediate quantization scale parameters. An audio inverse quantizer according to any one of claims 15 to 20.

22. the first-stage vector dequantizer (2220) is configured to take a single combined index as a first-stage dequantization result, and to process the single combined index to obtain the two or more result indexes.

22. An audio inverse quantizer as claimed in claim 21.

23. the first stage vector dequantizer (2220) is configured to retrieve a first result index of the two or more result indexes by determining a remainder from a division, and to retrieve a second result index of the two or more result indexes by determining an integer result from the division.

23. An audio inverse quantizer as claimed in claim 22.

24. 24. The audio inverse quantizer of claim 15, wherein the second stage vector inverse quantizer (2260) is an algebraic vector inverse quantizer, and each quantization index includes a base codebook index and a Voronoi extension index.

25. the first stage vector inverse quantizer (2220) or the combiner (2240) is configured to combine a first set of scale parameters and a second set of scale parameters from a quantization split in a frame of the multi-channel audio signal; the second stage vector dequantizer (2260) is configured to combine a first set of residual parameters and a second set of residual parameters from a split of residual parameters; the number of splits accommodated by the first stage vector inverse quantizer (2220) is the same as the number of other splits accommodated by the second stage vector inverse quantizer (2260); An audio inverse quantizer according to any one of claims 15 to 24.

26. the first stage vector inverse quantizer (2220) is configured to use a first quantization index having a first number of bits to generate the plurality of intermediate quantized audio information items; the second stage vector inverse quantizer (2260) is configured to use a second quantization index having a second number of bits as a quantization index to obtain the plurality of residual items, the second number of bits being less than or greater than the first number of bits; An audio inverse quantizer according to any one of claims 15 to 19.

27. 27. The audio inverse quantizer of claim 26, wherein the first number of bits is between 4 and 7, and the second number of bits is between 3 and 6.

28. the combiner (2240) is configured to attenuate the fourth plurality of side scale parameters for the second frame before further using or processing the fourth plurality of side scale parameters; 16. An audio inverse quantizer according to claim 15.

29. 1. A method for quantizing a plurality of audio information items, comprising the steps of: first stage vector quantization of said plurality of audio information items to determine a first stage vector quantization result and a plurality of intermediate quantization items corresponding to said first stage vector quantization result; calculating a plurality of residual items from said plurality of intermediate quantization items and said plurality of audio information items; second-stage vector quantization of the plurality of residual items to obtain a second-stage vector quantization result, the first-stage vector quantization result and the second-stage vector quantization result being a quantized representation of the plurality of audio information items; Including, said audio information item comprising, for a first frame of a multi-channel audio signal, a first plurality of scale parameters for a first channel of said multi-channel audio signal and a second plurality of scale parameters for a second channel of said multi-channel audio signal; the method includes applying the first stage vector quantizing and the second stage vector quantizing to the first plurality of scale parameters for the first channel of the first frame and the second plurality of scale parameters for the second channel of the first frame; said audio information item comprising, for a second frame of said multi-channel audio signal, a third plurality of mid-scale parameters and a fourth plurality of side-scale parameters; The method includes applying the first stage vector quantization and the second stage vector quantization to the third plurality of mid-scale parameters of the second frame and the fourth plurality of side-scale parameters of the second frame; applying the second stage vector quantization to the fourth plurality of side scale parameters of the second frame; not applying the first stage vector quantization to the fourth plurality of side scale parameters of the second frame. method.

30. 1. A method for dequantizing a plurality of quantized audio information items, comprising the steps of: - first-stage vector dequantization of first-stage vector quantization results contained in said quantized audio information items to obtain a plurality of intermediate quantized audio information items; - second stage vector dequantizing second stage vector quantization results contained in said quantized audio information items to obtain a plurality of residual items; combining the intermediate quantized audio information items and the residual items to obtain the dequantized audio information items. Including, the quantized audio information items comprise, for a first frame of a multi-channel audio signal, a first plurality of scale parameters for a first channel of the multi-channel audio signal and a second plurality of scale parameters for a second channel of the multi-channel audio signal; the method includes applying the first stage vector dequantization and the second stage vector dequantization to the first plurality of scale parameters for the first channel of the first frame and the second plurality of scale parameters for the second channel of the first frame; the quantized audio information items include a third plurality of mid-scale parameters and a fourth plurality of side-scale parameters for a second frame of the multi-channel audio signal; The method includes applying the first stage vector dequantizing and the second stage vector dequantizing to the third plurality of mid scale parameters of the second frame and the fourth plurality of side scale parameters of the second frame, applying the second stage vector dequantizing to the fourth plurality of side scale parameters of the second frame, and not applying the first stage vector dequantizing to the fourth plurality of side scale parameters of the second frame; Including, method.

31. 31. A computer program for carrying out the method according to claim 29 or the method according to claim 30 when the computer program is running on a computer or processor.

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