Patent Description:
Furthermore, the present invention can be applied in Joint Coding of the Stereo Spectral Noise Shaping Parameters
Spectral noise shaping shapes the quantization noise in the frequency domain such that the quantization noise is minimally perceived by the human ear and therefore, 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.

In this approach [<NUM>] [<NUM>], the MDCT spectrum is partitioned into a number of non-uniform scale factor bands. For example, at <NUM>, the MDCT has <NUM> coefficients and it is partitioned into <NUM> scale factor bands. In each band, a scale factor is used to scale the MDCT coefficients of that band. A scalar quantizer with constant step size is then employed to quantize the scaled MDCT coefficients. At the decoder-side, inverse scaling is performed in each band, shaping the quantization noise introduced by the scalar quantizer.

The <NUM> scale factors are encoded into the bitstream as side-information. It usually requires a significantly high number of bits for encoding the scale factors, due to the relatively high number of scale factors and the required high precision. This can become a problem at low bitrate and/or at low delay.

In an MDCT-based TCX, a transform-based audio codec used in the MPEG-D USAC [<NUM>] and 3GPP EVS [<NUM>] standards, spectral noise shaping is performed with the help of an LPC-based perceptual filters, similar perceptual filter as used in recent ACELP- based speech codecs (e.g. AMR-WB).

In this approach, a set of <NUM> Linear Prediction Coefficients (LPCs) is first estimated on a pre-emphasized input signal. The LPCs are then weighted and quantized. The frequency response of the weighted and quantized LPCs is then computed in <NUM> uniformly spaced bands. The MDCT coefficients are then scaled in each band using the computed 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 every <NUM> bands, shaping the quantization noise introduced by the scalar quantizer.

This approach has a clear advantage over the AAC approach: it requires the encoding of only <NUM> (LPC) + <NUM> (global-gain) parameters as side-information (as opposed to the <NUM> parameters in AAC). Moreover, <NUM> LPCs can be efficiently encoded with a small number of bits by employing an LSF representation and a vector quantizer. Consequently, the approach of MDCT-based TCX requires less side-information bits as the approach of AAC, which can make a significant difference at low bitrate and/or low delay.

An improved MDCT-based TCX system is published in [<NUM>]. In this new approach, the autocorrelation (for estimating the LPCs) is no more performed in the time-domain but it is instead computed in the MDCT domain using an inverse transform of the MDCT coefficient energies. This allows using a non-uniform frequency scale by simply grouping the MDCT coefficients into <NUM> non-uniform bands and computing the energy of each band. It also reduces the complexity required to compute the autocorrelation.

In an improved technique for spectral noise shaping as described in [<NUM>] and implemented in Low Complexity Communication Codec (LC3 / LC3plus), low bitrate without substantial loss of quality can be obtained by scaling, on the encoder-side, with a higher number of scale factors and by downsampling the scale parameters on the encoder-side into a second set of <NUM> scale parameters (SNS parameters). Thus, a low bitrate side information on the one hand and, nevertheless, a high-quality spectral processing of the audio signal spectrum due to fine scaling on the other hand are obtained.

In the thesis described in [<NUM>], a set of linear prediction coefficients are computed not only by considering the inter-frame prediction but also considering the prediction from one channel to another. The <NUM>-dimensional set of coefficients calculated are then quantized and encoded using similar techniques as for single channel LP, but without considering quantization of the residual in the context of the thesis. However, implementation described comes with high delay and significant complexity and therefore, it is rather unsuitable for a real-time application that requires low delay, e.g. for communication systems.

In a stereo system like the MDCT-based system that is described in [<NUM>], preprocessing of the discrete L R channel signals is performed in order to scale the spectra using frequency domain noise-shaping to the "whitened domain". Then, joint stereo processing is performed to quantize and code the whitened spectra in an optimal fashion.

The scaling parameters for the spectral noise shaping techniques described before are quantized encoded independently for each channel. This results in a double bitrate of side information needed to be sent to the decoder through the bitstream.

<CIT> discloses techniques and tools for representing, coding, and decoding scale factor information. During encoding of scale factors, an encoder uses one or more of flexible scale factor resolution selection, spatial prediction of scale factors, flexible prediction of scale factors, smoothing of noisy scale factor amplitudes, reordering of scale factor prediction residuals, and prediction of scale factor prediction residuals. During decoding, a decoder uses one or more of flexible scale factor resolution selection, spatial prediction of scale factors, flexible prediction of scale factors, reordering of scale factor prediction residuals, and prediction of scale factor prediction residuals.

It is an object of the present invention to provide an improved or more efficient coding/decoding concept.

This object is achieved by an audio decoder of claim <NUM>, an audio encoder of claim <NUM>, a method of decoding of claim <NUM>, a method of encoding of claim <NUM>, or a computer program of claim <NUM>.

The present invention is based on the finding that bitrate savings can be obtained for cases, where the L, R signals or, generally, two or more channels of a multi-channel signal are correlated. In such a case, the extracted parameters for both channels are rather similar.

Therefore, a joint quantization encoding of the parameters is applied which results in a significant saving of bitrate. This saving of bitrate can be used in several different directions. One direction can be to spend the saved bitrate on the coding of the core signal so that the overall perceptual quality of the stereo or multichannel signal is improved. Another direction is to reach a lower overall bitrate in a case where the coding of the core signal and, therefore, the overall perceptual quality is not improved, but is left at the same quality.

Independent claim <NUM> defines, in accordance with a first aspect, an audio encoder for encoding a multi-channel audio signal comprising two or more channels, comprising 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 for a second set of scale parameters for a second channel of the multi-channel audio signal. The audio encoder additionally comprises a signal processor for applying the first set of scale parameters to the first channel and for applying the second set of scale parameters to the second channel of the multi-channel audio signal. The signal processor additionally derives multi-channel audio data from the first and second channel data obtained by the application of the first and second sets of scale parameters, respectively. The audio encoder additionally has an encoded signal former for using the multi-channel audio data and the information on the first group of jointly encoded scale parameters and the information on the second group of jointly encoded scale parameters to obtain an encoded multi-channel audio signal.

Preferably, the scale parameter calculator is configured to be adaptive so that, for each frame or sub-frame of the multi-channel audio signal, a determination is made, whether jointly encoding scale parameters or separately encoding scale parameters is to be performed. In a further embodiment, this determination is based on a similarity analysis between the channels of the multi-channel audio signal under consideration. Particularly, the similarity analysis is done by calculating an energy of the jointly encoded parameters and, particularly, an energy of one set of scale parameters from the first group and the second group of jointly encoded scale parameters. Particularly, the scale parameter calculator calculates the first group as a sum between corresponding first and second scale parameters and calculates the second group as a difference between the first and second corresponding scale parameters. Particularly, the second group and, preferably, the scale parameters that represent the difference, are used for the determination of the similarity measure in order to decide, whether jointly encoding the scale parameters or separately encoding the scale parameters is to be performed. This situation can be signaled via a stereo or multi-channel flag.

Furthermore, it is preferred to specifically quantize the scale parameters with a two-stage quantization process. A first stage vector quantizer quantizes the plurality of scale parameters or, generally, audio information items to determination a first stage vector quantization result and to determinate a plurality of intermediate quantizer items corresponding to the first stage vector quantization result. Furthermore, the quantizer comprises a residual item determiner for calculating a plurality of residual items from the plurality of intermediate quantized 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, wherein the first stage vector quantization result and the second stage vector quantization result together represent the quantized representation of the plurality of audio information items which are, in one embodiment, the scale parameters. Particularly, the audio information items can either be jointly encoded scale parameters or separately encoded scale parameters. Furthermore, other audio information items can be any audio information items that are useful for vector quantization. Particularly, apart from scale parameters or scale factors as specific audio information items, other audio information items useful for the vector-quantized are spectral values such as MDCT or FFT lines. Even further audio information items that can 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 be it a spectral or a time envelope data representation.

In a preferred implementation, the residual item determiner calculates, for each residual item, a difference between corresponding audio information items such as a scale parameter and a corresponding intermediate quantized item such as a quantized scale parameter or scale factor. Furthermore, the residual item determiner is configured to amplify or weight, for each residual item, a difference between a corresponding audio information item and a corresponding intermediate quantized item so that the plurality of residual items are greater than the corresponding difference or to amplify or weigh the plurality of audio information items and/or the plurality of intermediate quantized items before calculating a difference between the amplified items to obtain the residual items. By this procedure, a useful control of the quantization error can be made. Particularly, when the second group of audio information items such as the different scale parameters are quite small, which is typically the case, when the first and the second channels are correlated to each other so that joint quantization has been determined, the residual items are typically quite small.

Therefore, when the residual items are amplified, the result of the quantization will comprise more values that are not quantized to <NUM> compared to a case, where this amplification has not been performed. Therefore, an amplification on the encoder or quantization side may be useful.

This is particularly the case when as in another preferred embodiment, the quantization of the jointly encoded second group of scale parameters, such as the difference scale parameters, is performed. Due to the fact that these side scale parameters are anyway small, a situation may arise that, without the amplification, most of the different scale parameters are quantized to <NUM> anyway. Therefore, in order to avoid this situation which might result in a loss of stereo impression and, therefore, in a loss of psychoacoustic quality, the amplification is performed so that only a small amount or almost no side scale parameters are quantized to <NUM>. This, of course, reduces the savings in bitrate. Due to this fact, however, the quantized residual data items are anyway only small, i.e., result in quantization indexes that represent small values and the bitrate increase is not too high, since quantization indexes for small values are encoded more efficiently than quantization indexes for higher values. This can even be enhanced by additionally performing an entropy coding operation that even more favors small quantization indexes with respect to bitrate over higher quantization indexes.

In another preferred embodiment, the first stage vector quantizer is a vector quantizer having a certain codebook and the second stage vector quantizer is an algebraic vector quantizer resulting, as a quantization index, in a codebook number, a vector index in a base codebook and a Voronoi index. Preferably, both the vector quantizer and the algebraic vector quantizer are configured to perform a split level vector quantization where both quantizers have the same split level procedure. Furthermore, the first and the second stage vector quantizers are configured in such a way that the number of bits and, therefore, the precision of the first stage vector quantizer result is greater than the number of bits or the 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 the 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 and the second stage vector quantizers are different from each other.

independent claim <NUM> defines 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, in accordance with the first aspect, the audio decoder comprises a scale parameter decoder for decoding the information on the jointly encoded scale parameters. Additionally, the audio decoder has a signal processor, where the scale parameter decoder is configured to combine a jointly encoded scale parameter of the first group and the jointly encoded scale parameter of the second group using different combination rules to obtain the scale parameters for the first set of scale parameters and the scale parameters for the second set of scale parameters that are then used by the signal processor.

In accordance with the further aspect of the present invention, an audio dequantizer is provided that comprises a first stage vector dequantizer, a second stage vector dequantizer and a combiner for combining the plurality of intermediate quantizer information items obtained by the first stage vector dequantizer and the plurality of residual items obtained from the second stage vector dequantizer to obtain a dequantized plurality of audio information items.

The first aspect of joint scale parameter coding can be combined with the second aspect related to the two stage vector quantization. On the other hand, the aspect of the two stage vector quantization can be applied to separately encoded scale parameters such as scale parameters for a left channel and a right channel or can be applied to the mid-scale parameters as another kind of audio information item. Thus, the second aspect of two-stage vector quantization can be applied independent from the first aspect or together with the first aspect.

Subsequently, preferred embodiments of the present invention are summarized.

In a stereo system where transform-based (MDCT) coding is used, the scaling parameters that are extracted from any of the techniques described in the introductory section for performing the frequency-domain noise shaping in the encoder side, need to be quantized and coded to be included as side-information to the bitstream. Then in the decoder side, scaling parameters are decoded and used to scale the spectrum of each channel to shape quantization noise in a manner that is minimally perceived.

Independent coding of spectral noise shaping parameters of the two channels: left and right can be applied.

Spectral noise shaping scaling parameters are coded adaptively either independently or jointly, depending on the degree of correlation between the two channels. In summary:.

In <FIG> an MDCT-stereo based encoder implementation is shown as described in detail in [<NUM>]. An essential part of the stereo system described in [<NUM>] is that the stereo processing is performed on the "whitened" spectra. Therefore, each channel undergoes a preprocessing, where for each frame, after windowing, the time domain block is transformed to the MDCT-domain, then Temporal Noise Shaping (TNS) is applied adaptively, either before or after the Spectral Noise Shaping (SNS) depending on the signal characteristics. After spectral noise shaping, joint stereo processing is performed, namely an adaptive band-wise M-S, L/R decision, to quantize and code the whitened spectra coefficients in an efficient manner. As a next step, stereo Intelligent Gap Filling (IGF) analysis is done and respective information bits are written to the bitstream. Finally, the processed coefficients are quantized and coded. Similar reference numbers as in <FIG> have been added. The calculation and processing of the scaling factors takes place in the blocks SNS between the two TNS blocks in <FIG>. The block window illustrates a windowing operation. The block MCLT stands for modified complex lapped transform. The block MDCT stands for modified discrete cosine transform. The block power spectrum stands for the calculation of a power spectrum. The block block switching decision stands for an analysis of the input signal to determine block lengths to be used for windowing. The block TNS stands for temporal noise shaping and this feature is performed either before or after the scaling of the spectrum in the block SNS.

In the MDCT-stereo codec implementation described in [<NUM>], at the encoder side preprocessing of the discrete L-R channels is performed in order to scale the spectra using frequency domain noise-shaping to the "whitened domain". Then, joint stereo processing is performed to quantize and code the whitened spectra in an optimal fashion.

At the decoder side, as depicted in <FIG> and described in [<NUM>], the encoded signal is decoded and inverse quantization and inverse stereo processing is performed. Then, the spectrum of each channel is "de-whitened" by the spectral noise shaping parameters that are retrieved from the bitstream. Similar reference numbers as in <FIG> have been added. The decoding and processing of the scale factors takes place in the blocks <NUM> in <FIG>. The blocks indicated in the figure are related to the blocks in the encoder in <FIG> 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.

The frequency-domain noise shaping (FDNS) applied in the system in [<NUM>] is here replaced with SNS as described in [<NUM>]. A block diagram of the processing path of SNS is shown in the block diagrams of <FIG> and <FIG> for the encoder and the decoder respectively.

Preferably, a low bitrate without substantial loss of quality can be obtained by scaling, on the encoder-side, with a higher number of scale factors and by downsampling the scale parameters on the encoder-side into a second set of scale parameters or scale factors, where the scale parameters in the second set that is then encoded and transmitted or stored via an output interface is lower than the first number of scale parameters. Thus, a fine scaling on the one hand and a low bitrate on the other hand is obtained on the encoder-side.

On the decoder-side, the transmitted small number of scale factors is 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 of the second set and, then, once again, a fine scaling using the higher number of scale parameters is performed on the decoder-side within a spectral processor to obtain a fine-scaled spectral representation.

Thus, a low bitrate on the one hand and, nevertheless, a high quality spectral processing of the audio signal spectrum on the other hand are obtained.

Spectral noise shaping as done in preferred embodiments is implemented using only a very low bitrate. Thus, this spectral noise shaping can be an essential tool even in a low bitrate transform-based audio codec. The spectral noise shaping shapes the quantization noise in the frequency domain such that the quantization noise is minimally perceived by the human ear and, therefore, the perceptual quality of the decoded output signal can be maximized.

Preferred embodiments rely on spectral parameters calculated from amplitude-related measures, such as energies of a spectral representation. Particularly, band-wise energies or, generally, band-wise amplitude-related measures are calculated as the basis for the scale parameters, where the bandwidths used in calculating the band-wise amplitude-related measures increase from lower to higher bands in order to approach the characteristic of the human hearing as far as possible. Preferably, the division of the spectral representation into bands is done in accordance with the well-known Bark scale.

In further embodiments, linear-domain scale parameters are calculated and are particularly calculated for the first set of scale parameters with the high number of scale parameters, and this high number of scale parameters is converted into a log-like domain. A log-like domain is generally a domain, in which small values are expanded and high values are compressed. Then, the downsampling or decimation operation of the scale parameters is done in the log-like domain that can be a logarithmic domain with the base <NUM>, or a logarithmic domain with the base <NUM>, where the latter is preferred for implementation purposes. The second set of scale factors is then calculated in the log-like domain and, preferably, a vector quantization of the second set of scale factors is performed, wherein the scale factors are in the log-like domain. Thus, the result of the vector quantization indicates log-like domain scale parameters. The second set of scale factors or scale parameters has, for example, a number of scale factors half of the number of scale factors of the first set, or even one third or yet even more preferably, one fourth. Then, the quantized small number of scale parameters in the second set of scale parameters is brought into the bitstream and is then transmitted from the encoder-side to the decoder-side or stored as an encoded audio signal together with a quantized spectrum that has also been processed using these parameters, where this processing additionally involves quantization using a global gain. Preferably, however, the encoder derives from these quantized log-like domain second scale factors once again a set of linear domain scale factors, which is the third set of scale factors, and the number of scale factors in the third set of scale factors is greater than the second number and is preferably even equal to the first number of scale factors in the first set of first scale factors. Then, on the encoder-side, these interpolated scale factors are used for processing the spectral representation, where the processed spectral representation is finally quantized and, in any way entropy-encoded, such as by Huffman-encoding, arithmetic encoding or vector-quantization-based encoding, etc..

In the decoder that receives an encoded signal having a low number of spectral parameters together with the encoded representation of the spectral representation, the low number of scale parameters is interpolated to a high number of scale parameters, i.e., to obtain a first set of scale parameters where a number of scale parameters of the scale factors of the second set of scale factors or scale parameters is smaller than the number of scale parameters of the first set, i.e., the set as calculated by the scale factor/parameter decoder. Then, a spectral processor located within the apparatus for decoding an encoded audio signal 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 that is preferably in the time domain.

Further embodiments result in additional advantages set forth below. In preferred embodiments, spectral noise shaping is performed with the help of <NUM> scaling parameters similar to the scale factors used in [<NUM>] or [<NUM>] or [<NUM>]. These parameters are obtained in the encoder by first computing the energy of the MDCT spectrum in <NUM> non-uniform bands (similar to the <NUM> non-uniform bands of prior art <NUM>), then by applying some processing to the <NUM> energies (smoothing, pre-emphasis, noise-floor, log-conversion), then by downsampling the <NUM> processed energies by a factor of <NUM> to obtain <NUM> parameters which are finally normalized and scaled. These <NUM> parameters are then quantized using vector quantization (using similar vector quantization as used in prior art <NUM>/<NUM>). The quantized parameters are then interpolated to obtain <NUM> interpolated scaling parameters. These <NUM> scaling parameters are then used to directly shape the MDCT spectrum in the <NUM> non-uniform bands. Similar to prior art <NUM> and <NUM>, the scaled MDCT coefficients are then quantized using a scalar quantizer with a step size controlled by a global gain.

In a further embodiment, the information on the jointly encoded scale parameters for one of the two groups such as the second group preferably related to the side scale parameters does not comprise quantization indices or other quantization bits but only information such as a flag or 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 an analysis or by 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 is used by the decoder to calculate the first and the second set of scale parameters only using the first group of jointly encoded scale parameters.

In a further embodiment, the second group of jointly encoded scale parameters is quantized only using the second quantization stage of the two stage quantizer, which preferably is a variable rate quantizer stage. In this case, it is assumed that the first stage results in all zero quantized values, so that only the second stage is effective. In an even further embodiment, only the first quantization stage of the two stage quantizer, which preferably is 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 the residual items are assumed to be zero or smaller than the smallest or first quantization step size of the second quantization stage.

Preferred embodiments of the present invention are subsequently discussed with respect to the accompanying drawings, in which:.

<FIG> illustrates 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 <NUM> and a signal processor <NUM>, <NUM>, <NUM> illustrated in <FIG> as a single item. The scale parameter decoder <NUM> receives the information on the jointly encoded first group and second group 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 the first channel representation of the multi-channel audio data and the second channel representation of the multi-channel audio data and applies the first set of scale parameters to a first channel representation derived from the multi-channel audio data and applies the second set of scale parameters to the second channel representation derived from the multi-channel audio data to obtain the first channel and the second channel of the decoded audio signal at the output of block <NUM>, <NUM>, <NUM> of <FIG>. Preferably, the jointly encoded scale parameters comprise information on the first group of jointly encoded scale parameters such as mid-scale parameters and information on a second group of jointly encoded scale parameters such as side scale parameters. Furthermore, the scale parameter decoder <NUM> is configured to combine a jointly encoded scale parameter of the first group and a jointly encoded scale parameter of the second group using a first combination rule to obtain a scale parameter of the first set of scale parameters and to combine the same both jointly encoded scale parameters of the first and second groups using a second combination rule which is different from the first combination rule to obtain a scale parameter of the second set of scale parameters. Thus, the scale parameter decoder <NUM> applies two different combination rules.

In a preferred embodiment, the two different combination rules are a plus or addition combination rule on the one hand and a subtraction or difference 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 useful as well depending on the representation of the corresponding scale parameters of the first group and the second group or of the first set and the second set of scale parameters.

<FIG> illustrates a corresponding audio encoder for encoding a multi-channel audio signal comprising two or more channels. The audio encoder comprises a scale parameter calculator <NUM>, a signal processor <NUM> and an encoded signal former <NUM>, <NUM>. The scale parameter calculator <NUM> 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 from a second set of scale parameters for a second channel of the multi-channel audio signal. Additionally, the signal processor is configured 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 for deriving encoded multi-channel audio data. The multi-channel audio data are derived from the scaled first and second channels and the multi-channel audio data are used by the encoded signal former <NUM>, <NUM> together with the information on the first and the second group of jointly encoded scale parameters to obtain the encoded multi-channel audio signal at the output of block <NUM> in <FIG>.

<FIG> illustrates a further implementation of the decoder of <FIG>. Particularly, the bitstream is input into the signal processor <NUM> that performs, typically, entropy decoding and inverse quantization together with intelligent gap filling procedures (IGF procedures) and inverse stereo processing of the scaled or whitened channels. The output of block <NUM> are scaled or whitened decoded left and right or, generally, several decoded channels of a multi-channel signal. The bitstream comprises side information bits for the scale parameters for left and right in the case of separate encoding and side information bits for scaled jointly encoded scale parameters illustrated as M, S scale parameters in <FIG>. This data is introduced into the scale parameter or scale factor decoder <NUM> that, at its output, generates the decoded left scale factors and the decoded right scale factors that are then applied in the shape spectrum block <NUM>, <NUM> to finally obtain a preferably MDCT spectrum for left and right that can then be converted into a time domain using a certain inverse MDCT operation.

The corresponding encoder-side implementation is given in <FIG> starts from an MDCT spectrum having a left and a right channel that are input into a spectrum shaper 120a, and the output of the spectrum shaper 120a is input into a processor 120b that, for example, performs a stereo processing, intelligent gap filling operations on an encoder side and corresponding quantization and (entropy) coding operations. Thus, blocks 120a, 120b together represent the signal processor <NUM> of <FIG>. Furthermore, for the purpose of the calculation of the scale factors which is performed in the block compute SNS (spectral noise shaping) scale factors 120b, an MDST spectrum is provided as well, and the MDST spectrum together with the MDCT spectrum is forwarded into a power spectrum calculator 110a. Alternatively, the power spectrum calculator 110a can operate directly on the input signal without an MDCT or MDST spectrum procedure. Another way would be to calculate the power spectrum from a DFT operation rather than an MDCT and an MDST operation, for example. Furthermore, the scale factors are calculated by the scale parameter calculator <NUM> that is illustrated in <FIG> as a block quantization encoding of scale factors. Particularly, block <NUM> outputs, dependent on the similarity between the first and the second channel, either separate encoded scale factors for left and right or jointly encoded scale factors for M and S. This is illustrated in <FIG> to the right of block <NUM>. Thus, in this implementation, block 110b calculates the scale factors for left and right and block <NUM> then determines, whether separate encoding, i.e., encoding for the left and right scale factors is better or worse than encoding of jointly encoded scale factors, i.e., M and S scale factors derived from the separate scale factors by the two different combination rules such as an addition on the one hand and a subtraction on the other hand.

The result of block <NUM> are side information bits for L, R or M, S that are, together with the result of block 120b, introduced into an output bitstream illustrated by <FIG>.

<FIG> illustrates a preferred implementation of the encoder of <FIG> or <FIG>. The first channel is input into a block 1100a that determines the separate scale parameters for the first channel, i.e., for channel L. Additionally, the second channel is input into block 1100b that determines the separate scale parameters for the second channel, i.e., for R. Then, the scale parameters for the left channel and the scale parameters for the right channel are correspondingly downsampled by a downsampler 130a for the first channel and a downsampler 130b for the second channel. The results are downsampled parameters (DL) for the left channel and downsampled parameters (DR) for the right channel.

Then, both these data DL and DR are input into a joint scale parameter determiner <NUM>. The joint scale parameter determiner <NUM> generates the first group of jointly encoded scale parameters such as mid or M scale parameters and a second group of jointly encoded scale parameters such as side or S scale parameters. Both groups are input in corresponding vector quantizers 140a, 140b to obtain quantized values that are then, in a final entropy encoder 140c and to be encoded to obtain the information on the jointly encoded scale parameters.

The entropy encoder 140c may be implemented to perform an arithmetic entropy encoding algorithm or an entropy encoding algorithm with a one-dimensional or with one or more dimensional Huffman code tables.

Another implementation of the encoder is illustrated in <FIG>, where the downsampling is not performed with the separate scale parameters such as with left and right as illustrated at 130a, 130b in <FIG>. Instead, the order of operations of the joint scale parameter determination and the subsequent downsampling by the corresponding downsamplers 130a, 130b is changed. Whether the implementation of <FIG> or <FIG> is used, depends on the certain implementation, where the implementation of <FIG> is preferred, since the joint scale parameter determination <NUM> is already performed on the downsampled scale parameters, i.e., the two different combination rules performed by the scale parameter calculator <NUM> are typically performed on a lower number of inputs compared to the case in <FIG>.

<FIG> illustrates the implementation of a decoder for decoding an encoded audio signal having multi-channel audio data comprising data for two or more audio channels and information on jointly encoded scale parameters. The decoder in <FIG>, however, is only part of the whole decoder of <FIG>, since only a part of the signal processor and, particularly, the corresponding channel scalers 212a, 212b are illustrated in <FIG>. With respect to the scale parameter decoder <NUM>, this element comprises an entropy decoder <NUM> reversing the procedure performed by corresponding block 140c in <FIG>. Furthermore, the entropy decoder outputs quantized jointly encoded scale parameters, such as quantized M scale parameters and quantized S scale parameters. The corresponding groups of scale parameters are input into dequantizers <NUM> and <NUM> in order to obtain dequantized values for M and S. These dequantized values are then input into a separate scale parameter determiner <NUM> that outputs scale parameters for left and right, i.e., separate scale parameters. These corresponding scale parameters are input into interpolators 222a, 222b to obtain interpolated scale parameters for left (IL) and interpolated scale parameters for right (IR). Both of these data are input into a channel scaler 212a and 212b, respectively. Additionally, the channel scalers correspondingly receive the first channel representation subsequent to the whole procedure done by block <NUM> in <FIG>, for example. Correspondingly, channel scaler 212b also obtains its corresponding second channel representation as output by block <NUM> in <FIG>. Then, a final channel scaling or "shape spectrum" as it is named in <FIG> takes place to obtain a shaped spectral channel for left and right that are illustrated as "MDCT spectrum" in <FIG>. Then, a final frequency domain to time domain conversion for each channel illustrated at 240a, 240b can be performed in order to finally obtain a decoded first channel and a decoded second channel of a multi-channel audio signal in a time domain representation.

Particularly, the scale parameter decoder <NUM> illustrated in the left portion of <FIG> can be included within an audio decoder as shown in <FIG> or as collectively shown in <FIG>, but can also be included as a local decoder within an encoder as will be shown with respect to <FIG> explicitly showing the local scale parameter decoder <NUM> at the output of the scale parameter encoder <NUM>.

<FIG> illustrates a further implementation where, with respect to <FIG>, the order of interpolation and scale parameter determination to determine the separate scale parameters is exchanged. Particularly, the interpolation takes place with the jointly encoded scale parameters M and S using interpolators 222a, 222b of <FIG>, and the interpolated jointly encoded scale parameters such as IM and IS are input into the separate scale parameter determiner <NUM>. Then, the output of block <NUM> are the upsampled scale parameters, i.e., the scale parameters for each of the, for example, <NUM> bands illustrated in <FIG>.

<FIG> illustrates a further preferred implementation of the encoder of <FIG>, <FIG> or <FIG>, <FIG>. The first channel and the second channel are both introduced into an optional time domain-to-frequency domain converter such as 100a, 100b of <FIG>. The spectral representation output by blocks 100a, 100b is input into a channel scaler 120a that individually scales the spectral representation for the left and the right channel. Thus, the channel scaler 120a performs a shape spectrum operation illustrated in 120a of <FIG>. The output of the channel scaler is input into a channel processor 120b of <FIG>, and the processed channels output of the block 120b are input into the encoded signal former <NUM>,<NUM> to obtain the encoded audio signal.

Furthermore, for the purpose of the determination of the separately or jointly encoded scale parameters, a similarity calculator <NUM> is provided that receives, as an input, the first channel and the second channel directly in the time domain. Alternatively, the similarity calculator can receive the first channel and the second channel at the output of the time domain-to-frequency domain converters 100a, 100b, i.e., the spectral representation.

Although it will be outlined with respect to <FIG> that the similarity between the two channels is calculated based on the second group of jointly encoded scale parameters, i.e., based on the side scale parameters, it is to be noted that this similarity can also be calculated based on the time domain or spectral domain channels directly without explicit calculation of the jointly encoded scale parameters. Alternatively, the similarity can also be determined based on the first group of jointly encoded scale parameters, i.e., based on the mid-scale parameters. Particularly, when the energy of the side scale parameters is lower than a threshold, then it is determined that jointly encoding can be performed. Analogously, the energy of the mid-scale parameters in a frame can also be measured, and determination for a joint encoding can be done when the energy of the mid-scale parameters is greater than another threshold, for example. Thus, many different ways for determining the similarity between the first channel and the second channel can be implemented in order to decide for joint coding of scale parameters or separate coding of scale parameters. Nevertheless, it is to be mentioned that the determination for joint or separate coding of scale parameters does not necessarily have to be identical to the determination of joint stereo coding for the channels, i.e., whether two channels are jointly coded using a mid/side representation or are separately coded in a L, R representation. The determination of joint encoding of the scale parameters is done independent on the determination of stereo processing for the actual channels, since the determination of any kind of stereo processing performed in block 120b in <FIG> is done after and subsequent to a scaling or shaping of the spectrum using scale factors for mid and side. Particularly, as illustrated in <FIG>, block <NUM> can determine a joint coding. Thus, as illustrated by the arrow in <FIG> pointing to block <NUM>, the scale factors for M and S can occur within this block. In case of the application of a local scale parameter decoder <NUM> within the encoder of <FIG>, then the actually used scale parameters for shaping the spectrum, although being scale parameters for left and scale parameters for right are nevertheless derived from the encoded and decoded scale parameters for mid and side.

With respect to <FIG>, a mode decider <NUM> is provided. The mode decider <NUM> receives the output of the similarity calculator <NUM> and decides for a separate coding of the scale parameters when the channels are not sufficiently similar. When, however, it is determined that the channels are similar, then a joint coding of the scale parameters is determined by block <NUM>, and the information, whether the separate or the change joint coding of the scale parameters is applied, is signaled by a corresponding side information or flag <NUM> illustrated in <FIG> that is provided from block <NUM> to the encoded signal former <NUM>, <NUM>. Furthermore, the encoder comprises the scale parameter encoder <NUM> that receives the scale parameters for the first channel and the scale parameters for the second channel and encodes the scale parameters either separately or jointly as controlled by the mode decider <NUM>. The scale parameter encoder <NUM> may, in one embodiment, output the scale parameters for the first and the second channel as indicated by the broken lines so that the channel scaler 120a performs a scaling with the corresponding first and second channel scale parameters. However, it is preferred to apply a local scale parameter decoder <NUM> within the encoder so that the channel scaling takes place with the locally encoded and decoded scale parameters so that the dequantized scale parameters are applied for a channel scaling in the encoder. This has the advantage that exactly the same situation takes place within the channel scaler in the encoder and the decoder at least with respect to the used scale parameters for channel scaling or spectrum shaping.

<FIG> illustrates a further preferred embodiment of the present invention with respect to the audio encoder. An MDCT spectrum calculator <NUM> is provided that can, for example, be a time domain to frequency domain converter applying an MDCT algorithm. Furthermore, a power spectrum calculator 110a is provided as illustrated in <FIG>. The separate scale parameters are calculated by a corresponding calculator <NUM>, and for the purpose of calculating the jointly encoded scale parameters, an addition block 1200a and a subtraction block 1200b. Then, for the purpose of determining the similarity, an energy calculation per frame with the side parameters, i.e., the second group of jointly encoded scale parameters is performed. In block <NUM>, a comparison to a threshold is performed and this block being similar to the mode decider <NUM> for the frame of <FIG> outputs the mode flag or stereo flag for the corresponding frame. Additionally, the information is given to the controllable encoder that performs a separate or joint coding in the current frame. To this end, the controllable encoder <NUM> receives the scale parameters calculated by a block <NUM>, i.e., the separate scale parameters and, additionally, receives the jointly encoded scale parameters, i.e., the ones determined by block 1200a and 1200b.

Block <NUM> preferably generates a zero flag for the frame, when block <NUM> determines that all side parameters of a frame are quantized to <NUM>. This result will occur when the first and the second channel are very close to each other and the differences between the channels and, therefore, the differences between the scale factors are so that these differences are smaller than the lowest quantization threshold applied by the quantizer included in block <NUM>. Block <NUM> outputs the information on the jointly encoded or separately encoded scale parameters for the corresponding frame.

<FIG> illustrates an audio quantizer for quantizing a plurality of audio information items. The audio quantizer comprises a first stage vector quantizer <NUM>, <NUM> for quantizing the plurality of audio information items such as scale factors or scale parameters or spectral values, etc. to determine a first stage vector quantization result <NUM>. Additionally, block <NUM>, <NUM> generates a plurality of intermediate quantized items corresponding to the first stage vector quantization result. The intermediate quantized items are, for example, the values associated with the first stage result. When the first stage result identifies a certain codebook with, for example, <NUM> certain (quantized) values, then the intermediate quantized items are the <NUM> values associated to the codebook vector index being the first stage result <NUM>. The intermediate quantized items and the audio information items at the input into the first stage vector quantizer <NUM>, <NUM> are input into a residual item determiner for calculating a plurality of residual items from the plurality of intermediate quantized items and the plurality of audio information items. This is e.g. done by calculating a difference for each item between the original item and the quantized item. The residual items are input into a second stage vector quantizer <NUM> for quantizing the plurality of residual items to obtain the second stage vector quantization result. Then, the first stage vector quantization result at the output of block <NUM>, <NUM> and the second stage result at the output of block <NUM> together represent the quantized representation of the plurality of audio information items that is encoded by an optional encoded signal former <NUM>, <NUM> that outputs the quantized audio information items that are, in the preferred embodiment, not only quantized but are additionally entropy encoded.

A corresponding audio dequantizer is illustrated in <FIG>. The audio dequantizer comprises a first stage vector dequantizer <NUM> for dequantizing a first stage quantization result included in the quantized plurality of audio information items to obtain a plurality of intermediate quantized audio information items. Furthermore, a second stage vector dequantizer <NUM> is provided and is configured for dequantizing a second stage vector quantization result included in the quantized plurality of audio information items to obtain a plurality of residual items. Both, the intermediate items from block <NUM> and the residual items from block <NUM> are combined by a combiner <NUM> for combining the plurality of intermediate quantized audio items and the plurality of residual items to obtain a dequantized plurality of audio information items. Particularly, the intermediate quantized items at the output of block <NUM> are separately encoded scale parameters such as for L and R or the first group of the jointly encoded scale parameters e.g. for M , and the residual items may represent the jointly encoded side scale parameters, for example, i.e., the second group of jointly encoded scale parameters.

<FIG> illustrates a preferred implementation of the first stage vector quantizer <NUM>, <NUM> of <FIG>. In step <NUM>, a vector quantization of a first subset of scale parameters is performed to obtain a first quantization index. In a step <NUM>, a vector quantization of a second subset of scale parameters is performed to obtain a second quantization index. Furthermore, dependent on the implementation, a vector quantization of a third subset of scale parameters is performed as illustrated in block <NUM> to obtain a third quantization index that is an optional index. The procedure in <FIG> is applied when there is a split level quantization. Exemplarily, the audio input signal is separated into <NUM> bands illustrated in <FIG>. These <NUM> bands are downsampled to <NUM> bands/scale factors, so that the whole band is covered by <NUM> scale factors. These <NUM> scale factors are quantized by the first stage vector quantizer <NUM>, <NUM> in a split-level mode illustrated in <FIG>. The first <NUM> scale factors of the <NUM> scale factors of <FIG> that are obtained by downsampling the original <NUM> scale factors are vector-quantized by step <NUM> and, therefore, represent the first subset of scale parameters. The remaining <NUM> scale parameters for the <NUM> upper bands represent the second subset of scale parameters that are vector-quantized in step <NUM>. Dependent on the implementation, a separation of the whole set of scale parameters or audio information items does not necessarily have to be done in exactly two subsets, but can also be done in three subsets or even more subsets.

Independent on how many splits are performed, the indexes for each level together represent the first stage result. As discussed with respect to <FIG>, these indexes can be combined via an index combiner in <FIG> to have a single first stage index. Alternatively, the first stage result can consist of the first index, and the second index and a potential third index and probably even more indexes that are not combined, but that are entropy encoded as they are.

In addition to the corresponding indexes forming the first stage result, step <NUM>, <NUM>, <NUM> also provide the intermediate scale parameters that are used in block <NUM> for the purpose of calculating the residual scale parameters for the frame. Hence, step <NUM> that is performed by, for example, block <NUM> of <FIG>, results in the residual scale parameters that are then processed by an (algebraic) vector quantization performed by step <NUM> in order to generate the second stage result. Thus, the first stage result and the second stage result are generated for the separate scale parameters L, the separate scale parameters R and the first group of joint scale parameters M. However, as illustrated in <FIG>, the (algebraic) vector quantization of the second group of jointly coded scale parameters or side scale parameters is only performed by step <NUM> that is in a preferred implementation identical to step <NUM> and is performed again by block <NUM> of <FIG>.

In a further embodiment, the information on the jointly encoded scale parameters for one of the two groups such as the second group preferably related to the side scale parameters does not comprise quantization indices or other quantization bits but only information such as a flag or 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 certain value such as a small value.

This information is determined by the encoder by an analysis or by 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 certain value scale parameters or by generating small random scale parameters all being e.g. smaller than the smallest or first quantization stage or is used by the decoder to calculate the first and the second set of scale parameters only using the first group of jointly encoded scale parameters. Hence, instead of performing stage <NUM> in <FIG>, only the all zero flag for the second group of jointly encoded scale parameters is written as the second stage result. The calculation in block <NUM> can be omitted as well in this case and can be replaced by a decider for deciding whether the all zero flag is to be activated and transmitted or not. This decider can be controlled by a user input indicating a skip of the coding of the S parameters altogether or a bitrate information or can actually perform an analysis of the residual items. Hence, for the frame having the all zero bit, the scale parameter decoder does not perform any combination but calculates the second set of scale parameters only using the first group of jointly encoded scale parameters such as by dividing the encoded scale parameters of the first group by two or by weighting using another predetermined value.

In a further embodiment, the second group of jointly encoded scale parameters is quantized only using the second quantization stage of the two stage quantizer, which preferably is a variable rate quantizer stage. In this case, it is assumed that the first stage results in all zero quantized values, so that only the second stage is effective. This case is illustrated in <FIG>.

In an even further embodiment, only the first quantization stage such as <NUM>, <NUM>, <NUM> of the two stage quantizer in <FIG>, which preferably is a fixed rate quantization stage, is applied and the second stage <NUM> is not used at all for a time portion or frame of the audio signal. This case corresponds to a situation, where all the residual items are assumed to be zero or smaller than the smallest or first quantization step size of the second quantization stage. Then, <FIG>, item <NUM> would correspond to items <NUM>, <NUM>, <NUM> of <FIG> and item <NUM> could be omitted as well and can be replaced by a decider for deciding that only the first stage quantization is used or not. This decider can be controlled by a user input or a bitrate information or can actually perform an analysis of the residual items to determine that the residual items are small enough so that the accuracy of the second group of jointly encoded scale parameters quantized by the single stage only is sufficient.

In a preferred implementation of the present invention that is additionally illustrated in <FIG>, the algebraic vector quantizer <NUM> additionally performs a split level calculation and, preferably, performs the same split level operation as is performed by the vector quantizer. Thus, the subsets of the residual values correspond, with respect to the band number, to the subset of scale parameters. For the case of having two split levels, i.e., for the first <NUM> downsampled bands of <FIG>, the algebraic vector quantizer <NUM> generates the first level result. Furthermore, the algebraic vector quantizer <NUM> generates a second level result for the upper <NUM> downsampled scale factors or scale parameters or, generally, audio information items.

Preferably, the algebraic vector quantizer <NUM> is implemented as the algebraic vector quantizer defined in section <NUM>. <NUM> of <NPL>) mentioned as reference (<NUM>) where, the result of the corresponding split multi-rate lattice vector quantization is a codebook number for each <NUM> items, a vector index in the base codebook and an <NUM>-dimensional Voronoi index. However, in case of only having a single codebook, the codebook number can be avoided and only the vector index in the base codebook and the corresponding n-dimensional Voronoi index is sufficient. Thus, these items which are item a, item b and item c or only item b and item c for each level for the algebraic vector quantization result represent the second stage quantization result.

Subsequently, reference is made to <FIG> illustrating a corresponding decoding operation matching with the encoding of <FIG>, <FIG> or the encoding of <FIG> in accordance with the first or the second aspect of the present invention or in accordance with both aspects.

In step <NUM> of <FIG>, 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 <NUM> of <FIG> indicates a true value. Then, a first stage decoding <NUM> and a second stage decoding <NUM> is performed in order to re-do the procedures done by the encoder of <FIG> and, particularly, by the algebraic vector quantizer <NUM> described with respect to <FIG> or described with respect to <FIG>. In step <NUM>, it is assumed that the side scale factors are all <NUM>. In step <NUM>, it is checked by means of the <NUM> flag value, whether there actually come non-zero quantized scale factors for the frame. In case the <NUM> flag value indicates that there are non-zero side scale factors for the frame, then the quantized side scale factors are retrieved and decoded using the second stage decoding <NUM> or performing block <NUM> of <FIG> only. In block <NUM>, the jointly encoded scale parameters are transformed back to the separately encoded scale parameters in order to then output the quantized left and right scale parameters that can then be used for inverse scaling of the spectrum in the decoder.

When the stereo mode flag value indicates a value of zero or when it is determined that a separate coding has been used within the frame, then only first stage decoding <NUM> and second stage decoding <NUM> is performed for the left and right scale factors and, since the left and right scale factors are already in the separately encoded representation, any transformation such as block <NUM> is not required. The process of efficiently coding and decoding the SNS scale factors that are needed for scaling the spectrum before the stereo processing at the encoder side and after the inverse stereo processing in the decoder side is described below to show a preferred implementation of the present invention as an exemplary pseudo code with comments. <IMG>
<IMG>.

Any sort of quantization e. uniform or non-uniform scalar quantization and entropy or arithmetic coding can be used to represent the parameters. In the described implementation, as can be seen in the algorithm description, a <NUM>-stage vector quantization scheme is implemented:.

Since the side signal for highly correlated channels can be considered small, using the e.g. reduced-scale <NUM>nd stage AVQ only is sufficient to represent the corresponding SNS parameters. By skipping the <NUM>st stage VQ for these signals, a significant complexity and bit saving for coding of the SNS parameters can be achieved.

A pseudo code description of each stage of quantization implemented is given below. First stage with <NUM>-split vector quantization using <NUM> bits for each split:
<IMG>
<IMG>
<IMG>.

The indices that are output from the coding process are finally packed to the bitstream and sent to the decoder.

The AVQ procedure disclosed above for the second stage is preferable implemented as outlined in EVS referring to is the High-Rate LPC (subclause <NUM>. <NUM>) in the MDCT-based TCX chapter. Specifically for the second-stage Algebraic vector quantizer used it is stated <NUM>. <NUM> Algebraic vector quantizer, and the algebraic VQ used for quantizing the refinement is described in subclause <NUM>. In an embodiment, one has, for each index, a set of codewords for the base codebook index and set of codewords for the Voronoi index, and all this is entropy coded and therefore of variable bit rate. Hence, the parameters of the AVQ in each sub-band j consist of the codebook number, the vector index in base codebook and the n- (such as <NUM>-) dimensional Voronoi index.

At the decoder end the indices are extracted from the bitstream and are used to decode and derive the quantized values of the scale factors. A pseudo code example of the procedure is given below.

The procedure of the <NUM>-stage decoding is described in detail in the pseudocode below. <IMG>
<IMG>.

The procedure of the <NUM>-stage decoding is described in detail in the pseudocode below.

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 pseudocode below:
<IMG>.

Regarding scaling or amplification/weighting of the residual on the encoder side and scaling or attenuation/ weighting on the decoder side, the weighting factors are not calculated separately for each value or split but a single weight or a small number of different weight (as an approximation to avoid complexity) are used to scale all the parameters. This scaling is a factor that determines the trade-off of e.g. coarse quantization (more quantizations to zero) bitrate savings and quantization precision (with respective spectral distortion), and can be predetermined in the encoder so that this predetermined value does not have to be transmitted to the decoder but can be fixedly set or initialized in the decoder to save transmission bits. Therefore, a higher scaling of the residual would require more bits but have minimal spectral distortion, while reducing the scale would save additional bits and if spectral distortion is kept in an acceptable range, that could serve as a means of additional bitrate saving.

<FIG> illustrates a comparison in the number of bits for both channels in line with a current prior art implementation (described as "independent" above), the new independent implementation in accordance with the second aspect of the present invention and for the new joint implementation in accordance with the first aspect of the present invention. <FIG> illustrates a histogram where the vertical axis represents the frequency of occurrence and the horizontal axis illustrates the bins of total number of bits for coding the parameters for both channels.

Subsequently, further preferred embodiments are illustrated where a specific emphasis is given to the calculation of the scale factors for each audio channel and where additionally specific emphasis is given to the specific application of downsampling and upsampling of the scale parameters, which is applied either before or subsequent to the calculation of the jointly encoded scale parameters as illustrated with respect to <FIG>, <FIG>.

<FIG> illustrates an apparatus for encoding an audio signal <NUM>. The audio signal <NUM> preferably is available in the time-domain, although other representations of the audio signal such as a prediction-domain or any other domain would principally also be useful. The apparatus comprises a converter <NUM>, a scale factor calculator <NUM>, a spectral processor <NUM>, a downsampler <NUM>, a scale factor encoder <NUM> and an output interface <NUM>. The converter <NUM> is configured for converting the audio signal <NUM> into a spectral representation. The scale factor calculator <NUM> is configured for calculating a first set of scale parameters or scale factors from the spectral representation. The other channel is received at block <NUM>, and the scale parameters from the other channels are received by block <NUM>.

Throughout the specification, the term "scale factor" or "scale parameter" is used in order to refer to the same parameter or value, i.e., a value or parameter that is, subsequent to some processing, used for weighting some kind of spectral values. This weighting, when performed in the linear domain is actually a multiplying operation with a scaling factor.

However, when the weighting is performed in a logarithmic domain, then the weighting operation with a scale factor is done by an actual addition or subtraction operation. Thus, in the terms of the present application, scaling does not only mean multiplying or dividing but also means, depending on the certain domain, addition or subtraction or, generally means each operation, by which the spectral value, for example, is weighted or modified using the scale factor or scale parameter.

The downsampler <NUM> is configured for downsampling the first set of scale parameters to obtain a second set of scale parameters, wherein a second number of the scale parameters in the second set of scale parameters is lower than a first number of scale parameters in the first set of scale parameters. This is also outlined in the box in <FIG> stating that the second number is lower than the first number. As illustrated in <FIG>, the scale factor encoder is configured for generating an encoded representation of the second set of scale factors, and this encoded representation is forwarded to the output interface <NUM>. Due to the fact that the second set of scale factors has a lower number of scale factors than the first set of scale factors, the bitrate for transmitting or storing the encoded representation of the second set of scale factors is lower compared to a situation, in which the downsampling of the scale factors performed in the downsampler <NUM> would not have been performed.

Furthermore, the spectral processor <NUM> is configured for processing the spectral representation output by the converter <NUM> in <FIG> using a third set of scale parameters, the third set of scale parameters or scale factors having a third number of scale factors being greater than the second number of scale factors, wherein the spectral processor <NUM> is configured to use, for the purpose of spectral processing the first set of scale factors as already available from block <NUM> via line <NUM>. Alternatively, the spectral processor <NUM> is configured to use the second set of scale factors as output by the downsampler <NUM> for the calculation of the third set of scale factors as illustrated by line <NUM>. In a further implementation, the spectral processor <NUM> uses the encoded representation output by the scale factor/parameter encoder <NUM> for the purpose of calculating the third set of scale factors as illustrated by line <NUM> in <FIG>. Preferably, the spectral processor <NUM> does not use the first set of scale factors, but uses either the second set of scale factors as calculated by the downsampler or even more preferably uses the encoded representation or, generally, the quantized second set of scale factors and, then, performs an interpolation operation to interpolate the quantized second set of spectral parameters to obtain the third set of scale parameters that has a higher number of scale parameters due to the interpolation operation.

Thus, the encoded representation of the second set of scale factors that is output by block <NUM> either comprises a codebook index for a preferably used scale parameter codebook or a set of corresponding codebook indices. In other embodiments, the encoded representation comprises the quantized scale parameters of quantized scale factors that are obtained, when the codebook index or the set of codebook indices or, generally, the encoded representation is input into a decoder-side vector decoder or any other decoder.

Preferably, the spectral processor <NUM> uses the same set of scale factors that is also available at 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.

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. Exemplarily, for example, one could derive <NUM> scale factors in block <NUM>, and one could then downsample the <NUM> scale factors to <NUM> scale factors for transmission. Then, one could perform an interpolation not necessarily to <NUM> scale factors, but to <NUM> scale factors in the spectral processor <NUM>. Alternatively, one could perform an interpolation to an even higher number such as more than <NUM> scale factors as the case may be, as long as the number of scale factors transmitted in the encoded output signal <NUM> is smaller than the number of scale factors calculated in block <NUM> or calculated and used in block <NUM> of <FIG>.

Preferably, the scale factor calculator <NUM> is configured to perform several operations illustrated in <FIG>. These operations refer to a calculation <NUM> of an amplitude-related measure per band, where the spectral representation for one channel is input into block <NUM>. The calculation for the other channel will take place in a similar manner. A preferred amplitude-related measure per band is the energy per band, but other amplitude-related measures can be used as well, for example, the summation of the magnitudes of the amplitudes per band or the summation of squared amplitudes which corresponds to the energy. However, apart from the power of <NUM> used for calculating the energy per band, other powers such as a power of <NUM> that would reflect the loudness of the signal could also be used and, even powers different from integer numbers such as powers of <NUM> or <NUM> can be used as well in order to calculate amplitude-related measures per band. Even powers less than <NUM> can be used as long as it is made sure that values processed by such powers are positive- valued.

A further operation performed by the scale factor calculator can be an inter-band smoothing <NUM>. This inter-band smoothing is preferably used to smooth out the possible instabilities that can appear in the vector of amplitude-related measures as obtained by step <NUM>. If one would not perform this smoothing, these instabilities would be amplified when converted to a log-domain later as illustrated at <NUM>, especially in spectral values where the energy is close to <NUM>. However, in other embodiments, inter-band smoothing is not performed.

A further preferred operation performed by the scale factor calculator <NUM> is the pre-emphasis operation <NUM>. This pre-emphasis operation has a similar purpose as a pre-emphasis operation used in an LPC-based perceptual filter of the MDCT-based TCX processing as discussed before with respect to the prior art. This procedure increases the amplitude of the shaped spectrum in the low-frequencies that results in a reduced quantization noise in the low-frequencies.

However, depending on the implementation, the pre-emphasis operation - as the other specific operations - does not necessarily have to be performed.

A further optional processing operation is the noise-floor addition processing <NUM>. This procedure improves the quality of signals containing very high spectral dynamics such as, for example, 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 cost of an increase of quantization noise in the valleys, where the quantization noise is anyway not perceptible due to masking properties of the human ear such as the absolute listening threshold, the pre-masking, the post-masking or the general masking threshold indicating that, typically, a quite low volume tone relatively close in frequency to a high volume tone is not perceptible at all, i.e., is fully masked or is only roughly perceived by the human hearing mechanism, so that this spectral contribution can be quantized quite coarsely.

The noise-floor addition operation <NUM>, however, does not necessarily have to be performed.

Furthermore, block <NUM> indicates a log-like domain conversion. Preferably, a transformation of an output of one of blocks <NUM>, <NUM>, <NUM>, <NUM> in <FIG> is performed in a log-like domain. A log-like domain is a domain, in which values close to <NUM> are expanded and high values are compressed. Preferably, the log domain is a domain with basis of <NUM>, but other log domains can be used as well. However, a log domain with the basis of <NUM> is better for an implementation on a fixed-point signal processor.

The output of the scale factor calculator <NUM> is a first set of scale factors.

As illustrated in <FIG>, each of the blocks <NUM> to <NUM> can be bridged, i.e., the output of block <NUM>, for example, could already be the first set of scale factors. However, all the processing operations and, particularly, the log-like domain conversion are preferred. Thus, one could even implement the scale factor calculator by only performing steps <NUM> and <NUM> without the procedures in steps <NUM> to <NUM>, for example. At the output of block <NUM>, a set of scale parameters for a channel (such as L) is obtained and a set of scale parameters for the other channel (such as R) can also be obtained by a similar calculation.

Thus, the scale factor calculator is configured for performing one or two or more of the procedures illustrated in <FIG> as indicated by the input/output lines connecting several blocks.

<FIG> illustrates a preferred implementation of the downsampler <NUM> of <FIG> again for a single channel. The data for the other channel is calculated in a similar way. Preferably, a low-pass filtering or, generally, a filtering with a certain window w(k) is performed in step <NUM>, and, then, a downsampling/decimation operation of the result of the filtering is performed. Due to the fact that low-pass filtering <NUM> and in preferred embodiments the downsampling/decimation operation <NUM> are both arithmetic operations, the filtering <NUM> and the downsampling <NUM> can be performed within a single operation as will be outlined later on. Preferably, the downsampling/decimation operation is performed in such a way that an overlap among 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 two decimated calculated parameters is performed. Thus, step <NUM> performs a low-pass filter on the vector of scale parameters before decimation. This low-pass filter has a similar effect as the spreading function used in psychoacoustic models. It reduces the quantization noise at the peaks, at the cost of an increase of quantization noise around the peaks where it is anyway perceptually masked at least to a higher degree with respect to quantization noise at the peaks.

Furthermore, the downsampler additionally performs a mean value removal <NUM> and an additional scaling step <NUM>. However, the low-pass filtering operation <NUM>, the mean value removal step <NUM> and the scaling step <NUM> are only optional steps. Thus, the downsampler illustrated in <FIG> or illustrated in <FIG> can be implemented to only perform step <NUM> or to perform two steps illustrated in <FIG> such as step <NUM> and one of the steps <NUM>, <NUM> and <NUM>. Alternatively, the downsampler can perform all four steps or only three steps out of the four steps illustrated in <FIG> as long as the downsampling/decimation operation <NUM> is performed.

As outlined in <FIG>, audio operations in <FIG> performed by the downsampler are performed in the log-like domain in order to obtain better results.

<FIG> illustrates a preferred implementation of the spectral processor. The spectral processor <NUM> included within the encoder of <FIG> comprises an interpolator <NUM> that receives the quantized second set of scale parameters for each channel or alternatively for a group of jointly encoded scale parameters and that outputs the third set of scale parameters for a channel of for a 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 comprises a linear domain converter <NUM>. Then, a spectral shaping is performed in block <NUM> using the linear scale parameters on the one hand and the spectral representation on the other hand that is obtained by the converter <NUM>. Preferably, a subsequent temporal noise shaping operation, i.e., a prediction over frequency is performed in order to obtain spectral residual values at the output of block <NUM>, while the TNS side information is forwarded to the output interface as indicated by arrow <NUM>.

Finally, the spectral processor <NUM>, 120b has at least one of a scalar quantizer/encoder that is configured for receiving a single global gain for the whole spectral representation, i.e., for a whole frame, and a stereo processing functionality and an IGF processing functionality, etc. Preferably, the global gain is derived depending on certain bitrate considerations. Thus, the global gain is set so that the encoded representation of the spectral representation generated by block <NUM>, 120b fulfils certain requirements such as a bitrate requirement, a quality requirement or both. The global gain can be iteratively calculated or can be calculated in a feed forward measure as the case may be. Generally, the global gain is used together with a quantizer and a high global gain typically results in a coarser quantization where a low global gain results in a finer quantization. Thus, in other words, a high global gain results in a higher quantization step size while a low global gain results in a smaller quantization step size when a fixed quantizer is obtained. However, other quantizers can be used as well together with the global gain functionality such as a quantizer that has some kind of compression functionality for high values, i.e., some kind of non-linear compression functionality so that, for example, the higher values are more compressed than lower values. The above dependency between the global gain and the quantization coarseness is valid, when the global gain is multiplied to the values before the quantization in the linear domain corresponding to an addition in the log domain. If, however, the global gain is applied by a division in the linear domain, or by a subtraction in the log domain, the dependency is the other way round. The same is true, when the "global gain" represents an inverse value.

Subsequently, preferred implementations of the individual procedures described with respect to <FIG> are given.

The energies per band EB(n) are computed as follows: <MAT> with X(k) are the MDCT coefficients, NB = <NUM> is the number of bands and Ind(n) are the band indices. The bands are non-uniform and follow the perceptually-relevant bark scale (smaller in low-frequencies, larger in high-frequencies).

The energy per band EB(b) is smoothed using <MAT>.

Remark: this step is mainly used to smooth the possible instabilities that can appear in the vector EB(b). If not smoothed, these instabilities are amplified when converted to log-domain (see step <NUM>), especially in the valleys where the energy is close to <NUM>.

The smoothed energy per band ES(b) is then pre-emphasized using <MAT> with gtilt controls the pre-emphasis tilt and depends on the sampling frequency. It is for example <NUM> at <NUM> and <NUM> at <NUM>. 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 <NUM>, it increases the amplitude of the shaped Spectrum in the low-frequencies, resulting in reduced quantization noise in the low-frequencies.

A noise floor at -40dB is added to EP(b) using <MAT> with the noise floor being calculated by <MAT>.

This step improves quality of signals containing very high spectral dynamics such as 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 cost of an increase of quantization noise in the valleys where it is anyway not perceptible.

A transformation into the logarithm domain is then performed using <MAT>.

The vector EL(b) is then downsampled by a factor of <NUM> using <MAT> With <MAT>.

This step applies a low-pass filter (w(k)) on the vector EL(b) before decimation. This low-pass filter has a similar effect as the spreading function used in psychoacoustic models: it reduces the quantization noise at the peaks, at the cost of an increase of quantization noise around the peaks where it is anyway perceptually masked.

The final scale factors are obtained after mean removal and scaling by a factor of <NUM> <MAT>.

Since the codec has an additional global-gain, the mean can be removed without any loss of information. Removing the mean also allows more efficient vector quantization. The scaling of <NUM> slightly compress the amplitude of the noise shaping curve. It has a similar perceptual effect as the spreading function mentioned in Step <NUM>: reduced quantization noise at the peaks and increased quantization noise in the valleys.

The scale factors are quantized using vector quantization, producing indices which are then packed into the bitstream and sent to the decoder, and quantized scale factors scfe(n).

The quantized scale factors scfQ(n) are interpolated using <MAT> <MAT> <MAT> <MAT> <MAT> <MAT> <MAT> <MAT> and transformed back into linear domain using <MAT>.

Interpolation is used to get a smooth noise shaping curve and thus to avoid any big amplitude jumps between adjacent bands.

The SNS scale factors gSNS(b) are applied on the MDCT frequency lines for each band separately in order to generate the shaped spectrum Xs(k) <MAT>.

<FIG> illustrates a preferred implementation of an apparatus for decoding an encoded audio signal <NUM> (a stereo signal encoded as L, R or M, S) comprising information on an encoded spectral representation and information on an encoded representation of a second set of scale parameters (separately of jointly encoded). The decoder comprises an input interface <NUM>, a spectrum decoder <NUM> (e.g. performing IGF processing or inverse stereo processing or dequantization processing), a scale factor/parameter decoder <NUM>, a spectral processor <NUM> (e.g. for R, L) and a converter <NUM> (e.g. for R, L). The input interface <NUM> is configured for receiving the encoded audio signal <NUM> and for extracting the encoded spectral representation that is forwarded to the spectrum decoder <NUM> and for extracting the encoded representation of the second set of scale factors that is forwarded to the scale factor decoder <NUM>. Furthermore, the spectrum decoder <NUM> is configured for decoding the encoded spectral representation to obtain a decoded spectral representation that is forwarded to the spectral processor <NUM>. The scale factor decoder <NUM> is configured for decoding the encoded second set of scale parameters to obtain a first set of scale parameters forwarded to the spectral processor <NUM>. The first set of scale factors has a number of scale factors or scale parameters that is greater than the number of scale factors or scale parameters in the second set. The spectral processor <NUM> is configured for processing the decoded spectral representation using the first set of scale parameters to obtain a scaled spectral representation. The scaled spectral representation is then converted by the converter <NUM> to finally obtain the decoded audio signal <NUM> being a stereo signal or a multichannel signal with more than two channels.

Preferably, the scale factor decoder <NUM> is configured to operate in substantially the same manner as has been discussed with respect to the spectral processor <NUM> of <FIG> relating to the calculation of the third set of scale factors or scale parameters as discussed in connection with blocks <NUM> or <NUM> and, particularly, with respect to blocks <NUM>, <NUM> of <FIG>. Particularly, the scale factor decoder is configured to perform the substantially same procedure for the interpolation and the transformation back into the linear domain as has been discussed before with respect to step <NUM>. Thus, as illustrated in <FIG>, the scale factor decoder <NUM> is configured for applying a decoder codebook <NUM> to the one or more indices per frame representing the encoded scale parameter representation. Then, an interpolation is performed in block <NUM> that is substantially the same interpolation as has been discussed with respect to block <NUM> in <FIG>. Then, a linear domain converter <NUM> is used that is substantially the same linear domain converter <NUM> as has been discussed with respect to <FIG>. However, in other implementations, blocks <NUM>, <NUM>, <NUM> can operate different from what has been discussed with respect to the corresponding blocks on the encoder-side.

Furthermore, the spectrum decoder <NUM> illustrated in <FIG> or <FIG> comprises a dequantizer/decoder block that receives, as an input, the encoded spectrum and that outputs a dequantized spectrum that is preferably dequantized using the global gain that is additionally transmitted from the encoder side to the decoder side within the encoded audio signal in an encoded form. The block <NUM> may also perform IGF processing or inverse stereo processing such as MS decoding. The dequantizer/decoder <NUM> can, for example, comprise an arithmetic or Huffman decoder functionality that receives, as an input, some kind of codes and that outputs quantization indices representing spectral values. Then, these quantization indices are input into a dequantizer together with the global gain and the output are dequantized spectral values that can then be subjected to a TNS processing such as an inverse prediction over frequency in a TNS decoder processing block <NUM> that, however, is optional. Particularly, the TNS decoder processing block additionally receives the TNS side information that has been generated by block <NUM> of <FIG> as indicated by line <NUM>. The output of the TNS decoder processing step <NUM> is input into a spectral shaping block <NUM> operating for each channel separately using the separate scale factors, where the first set of scale factors as calculated by the scale factor decoder are applied to the decoded spectral representation that can or cannot be TNS processed as the case may be, and the output is the scaled spectral representation for each channel that is then input into the converter <NUM> of <FIG>.

Further procedures of preferred embodiments of the decoder are discussed subsequently.

The vector quantizer indices produced in encoder step <NUM> are read from the bitstream and used to decode the quantized scale factors scfQ(n).

The SNS scale factors gSNS(b) are applied on the quantized MDCT frequency lines for each band separately in order to generate the decoded spectrum X̂(k) as outlined by the following code.

<FIG> and <FIG> illustrate a general encoder/decoder setup where <FIG> represents an implementation without TNS processing, while <FIG> illustrates an implementation that comprises TNS processing. Similar functionalities illustrated in <FIG> and <FIG> correspond to similar functionalities in the other figures when identical reference numerals are indicated. Particularly, as illustrated in <FIG>, the input signal <NUM> e.g. a stereo signal or a multichannel signal is input into a transform stage <NUM> and, subsequently, the spectral processing <NUM> is performed. Particularly, the spectral processing is reflected by an SNS encoder indicated by reference numerals <NUM>, <NUM>, <NUM>, <NUM> indicating that the block SNS encoder implements the functionalities indicated by these reference numerals. Subsequently to the SNS encoder block, a quantization encoding operation 120b, <NUM> is performed, and the encoded signal is input into the bitstream as indicated at <NUM> in <FIG>. The bitstream <NUM> then occurs at the decoder-side and subsequent to an inverse quantization and decoding illustrated by reference numeral <NUM>, the SNS decoder operation illustrated by blocks <NUM>, <NUM>, <NUM> of <FIG> are performed so that, in the end, subsequent to an inverse transform <NUM>, the decoded output signal <NUM> is obtained.

<FIG> illustrates a similar representation as in <FIG>, but it is indicated that, preferably, the TNS processing is performed subsequent to SNS processing on the encoder-side and, correspondingly, the TNS processing <NUM> is performed before the SNS processing <NUM> with respect to the processing sequence on the decoder-side.

Preferably the 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 does a time-domain shaping (as opposed to the frequency-domain shaping of SNS) as well. TNS is useful for signals containing sharp attacks and for speech signals.

TNS is usually applied (in AAC for example) between the transform and SNS. Preferably, however, it is preferred to apply TNS on the shaped spectrum. This avoids some artifacts that were produced by the TNS decoder when operating the codec at low bitrates.

<FIG> illustrates a preferred subdivision of the spectral coefficients or spectral lines as obtained by block <NUM> on the encoder-side into bands. Particularly, it is indicated that lower bands have a smaller number of spectral lines than higher bands.

Particularly, the x-axis in <FIG> corresponds to the index of bands and illustrates the preferred embodiment of <NUM> bands and the y-axis corresponds to the index of the spectral lines illustrating <NUM> spectral coefficients in one frame. Particularly, <FIG> illustrates exemplarily the situation of the super wide band (SWB) case where there is a sampling frequency of <NUM>.

For the wide band case, the situation with respect to the individual bands is so that one frame results in <NUM> spectral lines and the sampling frequency is <NUM> so that, for both cases, one frame has a length in time of <NUM> milliseconds.

<FIG> illustrates more details on the preferred downsampling performed in the downsampler <NUM> of <FIG> or the corresponding upsampling or interpolation as performed in the scale factor decoder <NUM> of <FIG> or as illustrated in block <NUM> of <FIG>.

Along the x-axis, the index for the bands <NUM> to <NUM> is given. Particularly, there are <NUM> bands going from <NUM> to <NUM>.

The <NUM> downsample points corresponding to scfQ(i) are illustrated as vertical lines <NUM>. Particularly, <FIG> illustrates how a certain grouping of scale parameters is performed to finally obtain the downsampled point <NUM>. Exemplarily, the first block of four bands consists of (<NUM>, <NUM>, <NUM>, <NUM>) and the middle point of this first block is at <NUM> indicated by item <NUM> at the index <NUM> along the x-axis.

Correspondingly, the second block of four bands is (<NUM>, <NUM>, <NUM>, <NUM>), and the middle point of the second block is <NUM>.

The windows <NUM> correspond to the windows w(k) discussed with respect to the step <NUM> downsampling described before. It can be seen that these windows are centered at the downsampled points and there is the overlap of one block to each side as discussed before.

The interpolation step <NUM> of <FIG> recovers the <NUM> bands from the <NUM> downsampled points. This is seen in <FIG> by computing the position of any of the lines <NUM> as a function of the two downsampled points indicated at <NUM> around a certain line <NUM>. The following example exemplifies that.

The position of the second band is calculated as a function of the two vertical lines around it (<NUM> and <NUM>) : <NUM>=<NUM>+<NUM>/8x(<NUM>-<NUM>).

Correspondingly, the position of the third band as a function of the two vertical lines <NUM> around it (<NUM> and <NUM>): <NUM>=<NUM>+<NUM>/8x(<NUM>-<NUM>).

A specific procedure is performed for the first two bands and the last two bands. For these bands, an interpolation cannot be performed, because there would not exist vertical lines or values corresponding to vertical lines <NUM> outside the range going from <NUM> to <NUM>. Thus, in order to address this issue, an extrapolation is performed as described with respect to step <NUM>: interpolation as outlined before for the two bands <NUM>, <NUM> on the one hand and <NUM> and <NUM> on the other hand.

Subsequently, a preferred implementation of the converter <NUM> of <FIG> on the one hand and the converter <NUM> of <FIG> on the other hand are discussed.

Particularly, <FIG> illustrates a schedule for indicating the framing performed on the encoder-side within converter <NUM>. <FIG> illustrates a preferred implementation of the converter <NUM> of <FIG> on the encoder-side and <FIG> illustrates a preferred implementation of the converter <NUM> on the decoder-side.

The converter <NUM> on the encoder-side is preferably implemented to perform a framing with overlapping frames such as a <NUM>% overlap so that frame <NUM> overlaps with frame <NUM> and frame <NUM> overlaps with frame <NUM> and frame <NUM>. However, other overlaps or a non-overlapping processing can be performed as well, but it is preferred to perform a <NUM>% overlap together with an MDCT algorithm. To this end, the converter <NUM> comprises an analysis window <NUM> and a subsequently-connected spectral converter <NUM> for performing an FFT processing, an MDCT processing or any other kind of time-to-spectrum conversion processing to obtain a sequence of frames corresponding to a sequence of spectral representations as input in <FIG> to the blocks subsequent to the converter <NUM>.

Correspondingly, the scaled spectral representation(s) are input into the converter <NUM> of <FIG>. Particularly, the converter comprises a time-converter <NUM> implementing an inverse FFT operation, an inverse MDCT operation or a corresponding spectrum-to-time conversion operation. The output is inserted into a synthesis window <NUM> and the output of the synthesis window <NUM> is input into an overlap-add processor <NUM> to perform an overlap-add operation in order to finally obtain the decoded audio signal. Particularly, the overlap-add processing in block <NUM>, for example, performs a sample-by-sample addition between corresponding samples of the second half of, for example, frame <NUM> and the first half of frame <NUM> so that the audio sampling values for the overlap between frame <NUM> and frame <NUM> as indicated by item <NUM> in <FIG> is obtained. Similar overlap-add operations in a sample-by-sample manner are performed to obtain the remaining audio sampling values of the decoded audio output signal.

It is to be mentioned here that all alternatives or aspects as discussed before and all aspects as defined by independent claims in the following claims can be used individually, i.e., without any other alternative or object than the contemplated alternative, object or independent claim. However, in other embodiments, two or more of the alternatives or the aspects or the independent claims can be combined with each other and, in other embodiments, all aspects, or alternatives and all independent claims can be combined to each other.

Although more aspects are described above, the attached claims indicate two different aspects, i.e., 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 Dequantizer, or Related Methods. These two aspects can be combined or used separately, as the case may be, and the inventions in accordance with these aspects are applicable to other application of audio processing different from the above described specific applications.

Furthermore, reference is made to the additional <FIG>, <FIG>, <FIG>, <FIG>, <FIG>, <FIG>, <FIG> illustrating the first aspect and <FIG> illustrating the second aspect and <FIG>, <FIG> illustrating the second aspect as applied within the first aspect.

An inventively encoded signal can be stored on a digital storage medium or a non-transitory storage medium or can be transmitted on a transmission medium such as a wireless transmission medium or a wired transmission medium such as the Internet.

Some embodiments comprise a data carrier having electronically
readable control signals, which are capable of cooperating with a programmable computer system, such that one of the methods described herein is performed.

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.

Claim 1:
Audio decoder for a 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, comprising:
a scale parameter decoder (<NUM>) for decoding the information on 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; and
a signal processor (<NUM>, <NUM>, <NUM>) 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,
wherein the jointly encoded scale parameters comprise information on a first group of jointly encoded scale parameters and information on a second group of jointly encoded scale parameters, and
wherein the scale parameter decoder (<NUM>) is configured to combine a jointly encoded scale parameter of the first group and a jointly encoded scale parameter of the second group using a first combination rule to obtain a scale parameter of the first set of scale parameters, and using a second combination rule being different from the first combination rule to obtain a scale parameter of the second set of scale parameters.