(19)
(11)EP 3 443 557 B1

(12)EUROPEAN PATENT SPECIFICATION

(45)Mention of the grant of the patent:
20.05.2020 Bulletin 2020/21

(21)Application number: 17715745.0

(22)Date of filing:  06.04.2017
(51)Int. Cl.: 
G10L 19/26  (2013.01)
(86)International application number:
PCT/EP2017/058238
(87)International publication number:
WO 2017/178329 (19.10.2017 Gazette  2017/42)

(54)

AUDIO ENCODER FOR ENCODING AN AUDIO SIGNAL, METHOD FOR ENCODING AN AUDIO SIGNAL AND COMPUTER PROGRAM UNDER CONSIDERATION OF A DETECTED PEAK SPECTRAL REGION IN AN UPPER FREQUENCY BAND

TONCODIERER ZUR CODIERUNG EINES TONSIGNALS, VERFAHREN ZUR CODIERUNG EINES TONSIGNALS UND COMPUTERPROGRAMM UNTER BERÜCKSICHTIGUNG EINES ERKANNTEN SPITZENSPEKTRALBEREICHS IN EINEM OBEREN FREQUENZBAND

CODEUR AUDIO SERVANT À CODER UN SIGNAL AUDIO, PROCÉDÉ DE CODAGE DE SIGNAL AUDIO ET PROGRAMME INFORMATIQUE PRENANT EN COMPTE UNE RÉGION SPECTRALE DE CRÊTE SUR UNE BANDE DE FRÉQUENCES SUPÉRIEURE


(84)Designated Contracting States:
AL AT BE BG CH CY CZ DE DK EE ES FI FR GB GR HR HU IE IS IT LI LT LU LV MC MK MT NL NO PL PT RO RS SE SI SK SM TR

(30)Priority: 12.04.2016 EP 16164951

(43)Date of publication of application:
20.02.2019 Bulletin 2019/08

(60)Divisional application:
20168799.3

(73)Proprietor: Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
80686 München (DE)

(72)Inventors:
  • MULTRUS, Markus
    90469 Nürnberg (DE)
  • NEUKAM, Christian
    90562 Kalchreuth (DE)
  • SCHNELL, Markus
    90409 Nürnberg (DE)
  • SCHUBERT, Benjamin
    90429 Nürnberg (DE)

(74)Representative: Zinkler, Franz et al
Schoppe, Zimmermann, Stöckeler Zinkler, Schenk & Partner mbB Patentanwälte Radlkoferstrasse 2
81373 München
81373 München (DE)


(56)References cited: : 
EP-A1- 2 980 794
WO-A1-2013/147668
WO-A1-2012/017621
  
      
    Note: Within nine months from the publication of the mention of the grant of the European patent, any person may give notice to the European Patent Office of opposition to the European patent granted. Notice of opposition shall be filed in a written reasoned statement. It shall not be deemed to have been filed until the opposition fee has been paid. (Art. 99(1) European Patent Convention).


    Description


    [0001] The present invention relates to audio encoding and, preferably, to a method, apparatus or computer program for controlling the quantization of spectral coefficients for the MDCT based TCX in the EVS codec. From the prior art EP2980794A1, an audio encoder using a frequency and time domain processing is known.

    [0002] A reference document for the EVS codec is 3GPP TS 24.445 V13.1.0 (2016-03), 3rd generation partnership project; Technical Specification Group Services and System Aspects; Codec for Enhanced Voice Services (EVS); Detailed algorithmic description (release 13).

    [0003] However, the present invention is additionally useful in other EVS versions as, for example, defined by other releases than release 13 and, additionally, the present invention is additionally useful in all other audio encoders different from EVS that, however, rely on a detector, a shaper and a quantizer and coder stage as defined, for example, in the claims.

    [0004] Additionally, it is to be noted that all embodiments defined not only by the independent but also defined by the dependent claims can be used separately from each other or together as outlined by the interdependencies of the claims or as discussed later on under preferred examples.

    [0005] The EVS Codec [1], as specified in 3GPP, is a modern hybrid-codec for narrow-band NB), wide-band (WB), super-wide-band (SWB) or full-band (FB) speech and audio content, which can switch between several coding approaches, based on signal classification:
    Fig. 1 illustrates a common processing and different coding schemes in EVS. Particularly, a common processing portion of the encoder in Fig. 1 comprises a signal resampling block 101, and a signal analysis block 102. The audio input signal is input at an audio signal input 103 into the common processing portion and, particularly, into the signal resampling block 101. The signal resampling block 101 additionally has a command line input for receiving command line parameters. The output of the common processing stage is input in different elements as can be seen in Fig. 1. Particularly, Fig. 1 comprises a linear prediction-based coding block (LP-based coding) 110, a frequency domain coding block 120 and an inactive signal coding/CNG block 130. Blocks 110, 120, 130 are connected to a bitstream multiplexer 140. Additionally, a switch 150 is provided for switching, depending on a classifier decision, the output of the common processing stage to either the LP-based coding block 110, the frequency domain coding block 120 or the inactive signal coding/CNG (comfort noise generation) block 130. Furthermore, the bitstream multiplexer 140 receives a classifier information, i.e., whether a certain current portion of the input signal input at block 103 and processed by the common processing portion is encoded using any of the blocks 110, 120, 130.
    • The LP-based (linear prediction based) coding, such as CELP coding, is primarily used for speech or speech-dominant content and generic audio content with high temporal fluctuation.
    • The Frequency Domain Coding is used for all other generic audio content, such as music or background noise.


    [0006] To provide maximum quality for low and medium bitrates, frequent switching between LP-based Coding and Frequency Domain Coding is performed, based on Signal Analysis in a Common Processing Module. To save on complexity, the codec was optimized to re-use elements of the signal analysis stage also in subsequent modules. For example: The Signal Analysis module features an LP analysis stage. The resulting LP-filter coefficients (LPC) and residual signal are firstly used for several signal analysis steps, such as the Voice Activity Detector (VAD) or speech/music classifier. Secondly, the LPC is also an elementary part of the LP-based Coding scheme and the Frequency Domain Coding scheme. To save on complexity, the LP analysis is performed at the internal sampling rate of the CELP coder (SRCELP).

    [0007] The CELP coder operates at either 12.8 or 16 kHz internal sampling-rate (SRCELP), and can thus represent signals up to 6.4 or 8 kHz audio bandwidth directly. For audio content exceeding this bandwidth at WB, SWB or FB, the audio content above CELP's frequency representation is coded by a bandwidth-extension mechanism.

    [0008] The MDCT-based TCX is a submode of the Frequency Domain Coding. Like for the LP-based coding approach, noise-shaping in TCX is performed based on an LP-filter. This LPC shaping is performed in the MDCT domain by applying gain factors computed from weighted quantized LP filter coefficients to the MDCT spectrum (decoder-side). On encoder-side, the inverse gain factors are applied before the rate loop. This is subsequently referred to as application of LPC shaping gains. The TCX operates on the input sampling rate (SRinp). This is exploited to code the full spectrum directly in the MDCT domain, without additional bandwidth extension. The input sampling rate SRinp, on which the MDCT transform is performed, can be higher than the CELP sampling rate SRCELP, for which LP coefficients are computed. Thus LPC shaping gains can only be computed for the part of the MDCT spectrum corresponding to the CELP frequency range (fCELP). For the remaining part of the spectrum (if any) the shaping gain of the highest frequency band is used.

    [0009] Fig. 2 illustrates on a high level the application of LPC shaping gains and for the MDCT based TCX.. Particularly, Fig. 2 illustrates a principle of noise-shaping and coding in the TCX or frequency domain coding block 120 of Fig. 1 on the encoder-side.

    [0010] Particularly, Fig. 2 illustrates a schematic block diagram of an encoder. The input signal 103 is input into the resampling block 201 in order to perform a resampling of the signal to the CELP sampling rate SRCELP, i.e., the sampling rate required by LP-based coding block 110 of Fig. 1. Furthermore, an LPC calculator 203 is provided that calculates LPC parameters and in block 205, an LPC-based weighting is performed in order to have the signal further processed by the LP-based coding block 110 in Fig. 1, i.e., the LPC residual signal that is encoded using the ACELP processor.

    [0011] Additionally, the input signal 103 is input, without any resampling, to a time-spectral converter 207 that is exemplarily illustrated as an MDCT transform. Furthermore, in block 209, the LPC parameters calculated by block 203 are applied after some calculations. Particularly, block 209 receives the LPC parameters calculated from block 203 via line 213 or alternatively or additionally from block 205 and then derives the MDCT or, generally, spectral domain weighting factors in order to apply the corresponding inverse LPC shaping gains. Then, in block 211, a general quantizer/encoder operation is performed that can, for example, be a rate loop that adjusts the global gain and, additionally, performs a quantization/coding of spectral coefficients, preferably using arithmetic coding as illustrated in the well-known EVS encoder specification to finally obtain the bitstream.

    [0012] In contrast to the CELP coding approach, which combines a core-coder at SRCELP and a bandwidth-extension mechanism running at a higher sampling rate, the MDCT-based coding approaches directly operate on the input sampling rate SRinp and code the content of the full spectrum in the MDCT domain.

    [0013] The MDCT-based TCX codes up to 16 kHz audio content at low bitrates, such as 9.6 or 13.2 kbit/s SWB. Since at such low bitrates only a small subset of the spectral coefficients can be coded directly by means of the arithmetic coder, the resulting gaps (regions of zero values) in the spectrum are concealed by two mechanisms:
    • Noise Filling, which inserts random noise in the decoded spectrum. The energy of the noise is controlled by a gain factor, which transmitted in the bitstream.
    • Intelligent Gap Filling (IGF), which inserts signal portions from lower frequencies parts of the spectrum. The characteristics of these inserted frequency-portions are controlled by parameters, which are transmitted in the bitstream.


    [0014] The Noise Filling is used for lower frequency portions up to the highest frequency, which can be controlled by the transmitted LPC (fCELP). Above this frequency, the IGF tool is used, which provides other mechanisms to control the level of the inserted frequency portions.

    [0015] There are two mechanisms for the decision on which spectral coefficients survive the encoding procedure, or which will be replaced by noise filling or IGF:
    1. 1) Rate loop
      After the application of inverse LPC shaping gains, a rate loop is applied. For this, a global gain is estimated. Subsequently, the spectral coefficients are quantized, and the quantized spectral coefficients are coded with the arithmetic coder. Based on the real or an estimated bit-demand of the arithmetic coder and the quantization error, the global gain is increased or decreased. This impacts the precision of the quantizer. The lower the precision, the more spectral coefficients are quantized to zero. Applying the inverse LPC shaping gains using a weighted LPC before the rate loop assures that the perceptually relevant lines survive by a significantly higher probability than perceptually irrelevant content.
    2. 2) IGF Tonal mask
      Above fCELP, where the no LPC is available, a different mechanism to identify the perceptually relevant spectral components is used: Line-wise energy is compared to the average energy in the IGF region. Predominant spectral lines, which correspond to perceptually relevant signal portions, are kept, all other lines are set to zero. The MDCT spectrum, which was preprocessed with the IGF Tonal mask is subsequently fed into the Rate loop.


    [0016] The weighted LPC follows the spectral envelope of the signal. By applying the inverse LPC shaping gains using the weighted LPC a perceptual whitening of the spectrum is performed. This significantly reduces the dynamics of the MDCT spectrum before the coding-loop, and thus also controls the bit-distribution among the MDCT spectral coefficients in the coding-loop.

    [0017] As explained above, the weighted LPC is not available for frequencies above fCELP. For these MDCT coefficients, the shaping gain of the highest frequency band below fCELP is applied. This works well in cases where the shaping gain of the highest frequency band below fCELP roughly corresponds to the energy of the coefficients above fCELP, which is often the case due to the spectral tilt, and which can be observed in most audio signals. Hence, this procedure is advantageous, since the shaping information for the upper band need not be calculated or transmitted.

    [0018] However, in case there are strong spectral components above fCELP and the shaping gain of the highest frequency band below fCELP is very low, this results in a mismatch. This mismatch heavily impacts the work or the rate loop, which focuses on the spectral coefficients having the highest amplitude. This will at low bitrates zero out the remaining signal components, especially in the low-band, and produces perceptually bad quality.

    [0019] Figures 3-6 illustrate the problem. Figure 3 shows the absolute MDCT spectrum before the application of the inverse LPC shaping gains, Figure 4 the corresponding LPC shaping gains. There are strong peaks above fCELP visible, which are in the same order of magnitude as the highest peaks below fCELP. The spectral components above fCELP are a result of the preprocessing using the IGF tonal mask. Figure 5 shows the absolute MDCT spectrum after applying the inverse LPC gains, still before quantization. Now the peaks above fCELP significantly exceed the peaks below fCELP, with the effect that the rate-loop will primarily focus on these peaks. Figure 6 shows the result of the rate loop at low bitrates: All spectral components except the peaks above fCELP were quantized to 0. This results in a perceptually very poor result after the complete decoding process, since the psychoacoustically very relevant signal portions at low frequencies are missing completely.

    [0020] Fig. 3 illustrates an MDCT spectrum of a critical frame before the application of inverse LPC shaping gains.

    [0021] Fig. 4 illustrates LPC shaping gains as applied. On the encoder-side, the spectrum is multiplied with the inverse gain. The last gain value is used for all MDCT coefficients above fCELP. Fig. 4 indicates fCELP at the right border.

    [0022] Fig. 5 illustrates an MDCT spectrum of a critical frame after application of inverse LPC shaping gains. The high peaks above fCELP are clearly visible.

    [0023] Fig. 6 illustrates an MDCT spectrum of a critical frame after quantization. The displayed spectrum includes the application of the global gain, but without the LPC shaping gains. It can be seen that all spectral coefficients except the peak above fCELP are quantized to 0.

    [0024] It is an object of the present invention to provide an improved audio encoding concept.

    [0025] This object is achieved by an audio encoder of claim 1, a method for encoding an audio signal of claim 25 or a computer program of claim 26.

    [0026] The present invention is based on the finding that such prior art problems can be addressed by preprocessing the audio signal to be encoded depending on a specific characteristic of the quantizer and coder stage included in the audio encoder. To this end, a peak spectral region in an upper frequency band of the audio signal is detected. Then, a shaper for shaping the lower frequency band using shaping information for the lower band and for shaping the upper frequency band using at least a portion of the shaping information for the lower band is used. Particularly, the shaper is additionally configured to attenuate spectral values in a detected peak spectral region, i.e., in a peak spectral region detected by the detector in the upper frequency band of the audio signal. Then, the shaped lower frequency band and the attenuated upper frequency band are quantized and entropy-encoded.

    [0027] Due to the fact that the upper frequency band has been attenuated selectively, i.e., within the detected peak spectral region, this detected peak spectral region cannot fully dominate the behavior of the quantizer and coder stage anymore.

    [0028] Instead, due to the fact that an attenuation has been formed in the upper frequency band of the audio signal, the overall perceptual quality of the result of the encoding operation is improved. Particularly at low bitrates, where a quite low bitrate is a main target of the quantizer and coder stage, high spectral peaks in the upper frequency band would consume all the bits required by the quantizer and coder stage, since the coder would be guided by the high upper frequency portions and would, therefore, use most of the available bits in these portions. This automatically results in a situation where any bits for perceptually more important lower frequency ranges are not available anymore. Thus, such a procedure would result in a signal only having encoded high frequency portions while the lower frequency portions are not coded at all or are only encoded very coarsely. However, it has been found that such a procedure is less perceptually pleasant compared to a situation, where such a problematic situation with predominant high spectral regions is detected and the peaks in the higher frequency range are attenuated before performing the encoder procedure comprising a quantizer and a entropy encoder stage.

    [0029] Preferably, the peak spectral region is detected in the upper frequency band of an MDCT spectral. However, other time-spectral converters can be used as well such as a filterbank, a QMF filter bank, a DFT, an FFT or any other time-frequency conversion.

    [0030] Furthermore, the present invention is useful in that, for the upper frequency band, it is not required to calculate shaping information. Instead, a shaping information originally calculated for the lower frequency band is used for shaping the upper frequency band. Thus, the present invention provides a computationally very efficient encoder since a low band shaping information can also be used for shaping the high band, since problems that might result from such a situation, i.e., high spectral values in the upper frequency band are addressed by the additional attenuation additionally applied by the shaper in addition to the straightforward shaping typically based on the spectral envelope of the low band signal that can, for example, be characterized by a LPC parameters for the low band signal. But the spectral envelope can also be represented by any other corresponding measure that is usable for performing a shaping in the spectral domain.

    [0031] The quantizer and coder stage performs a quantizing and coding operation on the shaped signal, i.e., on the shaped low band signal and on the shaped high band signal, but the shaped high band signal additionally has received the additional attenuation.

    [0032] Although the attenuation of the high band in the detected peak spectral region is a preprocessing operation that cannot be recovered by the decoder anymore, the result of the decoder is nevertheless more pleasant compared to a situation, where the additional attenuation is not applied, since the attenuation results in the fact that bits are remaining for the perceptually more important lower frequency band. Thus, in problematic situations where a high spectral region with peaks would dominate the whole coding result, the present invention provides for an additional attenuation of such peaks so that, in the end, the encoder "sees" a signal having attenuated high frequency portions and, therefore, the encoded signal still has useful and perceptually pleasant low frequency information. The "sacrifice" with respect to the high spectral band is not or almost not noticeable by listeners, since listeners, generally, do not have a clear picture of the high frequency content of a signal but have, to a much higher probability, an expectation regarding the low frequency content. In other words, a signal that has very low level low frequency content but a significant high level frequency content is a signal that is typically perceived to be unnatural.

    [0033] Preferred embodiments of the invention comprise a linear prediction analyzer for deriving linear prediction coefficients for a time frame and these linear prediction coefficients represent the shaping information or the shaping information is derived from those linear prediction coefficients.

    [0034] In a further embodiment, several shaping factors are calculated for several subbands of the lower frequency band, and for the weighting in the higher frequency band, the shaping factor calculated for the highest subband of the low frequency band is used.

    [0035] In a further embodiment, the detector determines a peak spectral region in the upper frequency band when at least one of a group of conditions is true, where the group of conditions comprises at least a low frequency band amplitude condition, a peak distance condition and a peak amplitude condition. Even more preferably, a peak spectral region is only detected when two conditions are true at the same time and even more preferably, a peak spectral region is only detected when all three conditions are true.

    [0036] In a further embodiment, the detector determines several values used for examining the conditions either before or after the shaping operation with or without the additional attenuation.

    [0037] In an embodiment, the shaper additionally attenuates the spectral values using an attenuation factor, where this attenuation factor is derived from a maximum spectral amplitude in the lower frequency band multiplied by a predetermined number being greater than or equal to 1 and divided by the maximum spectral amplitude in the upper frequency band.

    [0038] Furthermore, the specific way, as to how the additional attenuation is applied, can be done in several different ways. One way is that the shaper firstly performs the weighting information using at least a portion of the shaping information for the lower frequency band in order to shape the spectral values in the detected peak spectral region. Then, a subsequent weighting operation is performed using the attenuation information.

    [0039] An alternative procedure is to firstly apply a weighting operation using the attenuation information and to then perform a subsequent weighting using a weighting information corresponding to the at least the portion of the shaping information for the lower frequency band. A further alternative is to apply a single weighting information using a combined weighting information that is derived from the attenuation on the one hand and the portion of the shaping information for the lower frequency band on the other hand.

    [0040] In a situation where the weighting is performed using a multiplication, the attenuation information is an attenuation factor and the shaping information is a shaping factor and the actual combined weighting information is a weighting factor, i.e., a single weighting factor for the single weighting information, where this single weighting factor is derived by multiplying the attenuation information and the shaping information for the lower band. Thus, it becomes clear that the shaper can be implemented in many different ways, but, nevertheless, the result is a shaping of the high frequency band using shaping information of the lower band and an additional attenuation.

    [0041] In an embodiment, the quantizer and coder stage comprises a rate loop processor for estimating a quantizer characteristic so that the predetermined bitrate of an entropy encoded audio signal is obtained. In an embodiment, this quantizer characteristic is a global gain, i.e., a gain value applied to the whole frequency range, i.e., applied to all the spectral values that are to be quantized and encoded. When it appears that the required bitrate is lower than a bitrate obtained using a certain global gain, then the global gain is increased and it is determined whether the actual bitrate is now in line with the requirement, i.e., is now smaller than or equal to the required bitrate. This procedure is performed, when the global gain is used in the encoder before the quantization in such a way the spectral values are divided by the global gain. When, however, the global gain is used differently, i.e., by multiplying the spectral values by the global gain before performing the quantization, then the global gain is decreased when an actual bitrate is too high, or the global gain can be increased when the actual bitrate is lower than admissible.

    [0042] However, other encoder stage characteristics can be used as well in a certain rate loop condition. One way would, for example, be a frequency-selective gain. A further procedure would be to adjust the band width of the audio signal depending on the required bitrate. Generally, different quantizer characteristics can be influenced so that, in the end, a bit rate is obtained that is in line with the required (typically low) bitrate.

    [0043] Preferably, this procedure is particularly well suited for being combined with intelligent gap filling processing (IGF processing). In this procedure, a tonal mask processor is applied for determining, in the upper frequency band, a first group of spectral values to be quantized and entropy encoded and a second group of spectral values to be parametrically encoded by the gap-filling procedure. The tonal mask processor sets the second group of spectral values to 0 values so that these values do not consume many bits in the quantizer/encoder stage. On the other hand, it appears that typically values belonging to the first group of spectral values that are to be quantized and entropy coded are the values in the peak spectral region that, under certain circumstances, can be detected and additionally attenuated in case of a problematic situation for the quantizer/encoder stage. Therefore, the combination of a tonal mask processor within an intelligent gap-filling framework with the additional attenuation of detected peak spectral regions results in a very efficient encoder procedure which is, additionally, backward-compatible and, nevertheless, results in a good perceptual quality even at very low bitrates.

    [0044] Embodiments are advantageous over potential solutions to deal with this problem that include methods to extend the frequency range of the LPC or other means to better fit the gains applied to frequencies above FCELP to the actual MDCT spectral coefficients. This procedure, however, destroys backward compatibility, when a codec is already deployed in the market, and the previously described methods would break interoperability to existing implementations.

    [0045] Subsequently, preferred embodiments of the present invention are illustrated with respect to the accompanying drawings, in which:
    Fig. 1
    illustrates a common processing and different coding schemes in EVS;
    Fig. 2
    illustrates a principle of noise-shaping and coding in the TCX on the encoder-side;
    Fig. 3
    illustrates an MDCT spectrum of a critical frame before the application of inverse LPC shaping gains;
    Fig. 4
    illustrates the situation of Fig. 3, but with the LPC shaping gains applied;
    Fig. 5
    illustrates an MDCT spectrum of a critical frame after the application of inverse LPC shaping gains, where the high peaks above fCELP are clearly visible;
    Fig. 6
    illustrates an MDCT spectrum of a critical frame after quantization only having high pass information and not having any low pass information;
    Fig. 7
    illustrates an MDCT spectrum of a critical frame after the application of inverse LPC shaping gains and the inventive encoder-side pre-processing;
    Fig. 8
    illustrates a preferred embodiment of an audio encoder for encoding an audio signal;
    Fig. 9
    illustrates the situation for the calculation of different shaping information for different frequency bands and the usage of the lower band shaping information for the higher band;
    Fig, 10
    illustrates a preferred embodiment of an audio encoder;
    Fig. 11
    illustrates a flow chart for illustrating the functionality of the detector for detecting the peak spectral region;
    Fig. 12
    illustrates a preferred implementation of the implementation of the low band amplitude condition;
    Fig. 13
    illustrates a preferred embodiment of the implementation of the peak distance condition;
    Fig. 14
    illustrates a preferred implementation of the implementation of the peak amplitude condition;
    Fig. 15a
    illustrates a preferred implementation of the quantizer and coder stage;
    Fig. 15b
    illustrates a flow chart for illustrating the operation of the quantizer and coder stage as a rate loop processor;
    Fig. 16
    illustrates a determination procedure for determining the attenuation factor in a preferred embodiment; and
    Fig. 17
    illustrates a preferred implementation for applying the low band shaping information to the upper frequency band and the additional attenuation of the shaped spectral values in two subsequent steps.


    [0046] Fig. 8 illustrates a preferred embodiment of an audio encoder for encoding an audio signal 403 having a lower frequency band and an upper frequency band. The audio encoder comprises a detector 802 for detecting a peak spectral region in the upper frequency band of the audio signal 103. Furthermore, the audio encoder comprises a shaper 804 for shaping the lower frequency band using shaping information for the lower band and for shaping the upper frequency band using at least a portion of the shaping information for the lower frequency band. Additionally, the shaper is configured to additionally attenuate spectral values in the detected peak spectral region in the upper frequency band.

    [0047] Thus, the shaper 804 performs a kind of "single shaping" in the low-band using the shaping information for the low-band. Furthermore, the shaper additionally performs a kind of a "single" shaping in the high-band using the shaping information for the low-band and typically, the highest frequency low-band. This "single" shaping is performed in some embodiments in the high-band where no peak spectral region has been detected by the detector 802. Furthermore, for the peak spectral region within the high-band, a kind of a "double" shaping is performed, i.e., the shaping information from the low-band is applied to the peak spectral region and, additionally, the additional attenuation is applied to the peak spectral region.

    [0048] The result of the shaper 804 is a shaped signal 805. The shaped signal is a shaped lower frequency band and a shaped upper frequency band, where the shaped upper frequency band comprises the peak spectral region. This shaped signal 805 is forwarded to a quantizer and coder stage 806 for quantizing the shaped lower frequency band and the shaped upper frequency band including the peak spectral region and for entropy coding the quantized spectral values from the shaped lower frequency band and the shaped upper frequency band comprising the peak spectral region again to obtain the encoded audio signal 814.

    [0049] Preferably, the audio encoder comprises a linear prediction coding analyzer 808 for deriving linear prediction coefficients for a time frame of the audio signal by analyzing a block of audio samples in the time frame. Preferably, these audio samples are band-limited to the lower frequency band.

    [0050] Additionally, the shaper 804 is configured to shape the lower frequency band using the linear prediction coefficients as the shaping information as illustrated at 812 in Fig. 8. Additionally, the shaper 804 is configured to use at least the portion of the linear prediction coefficients derived from the block of audio samples band-limited to the lower frequency band for shaping the upper frequency band in the time frame of the audio signal.

    [0051] As illustrated in Fig. 9, the lower frequency band is preferably subdivided into a plurality of subbands such as, exemplarily four subbands SB1, SB2, SB3 and SB4. Additionally, as schematically illustrated, the subband width increases from lower to higher subbands, i.e., the subband SB4 is broader in frequency than the subband SB1. In other embodiments, however, bands having an equal bandwidth can be used as well.

    [0052] The subbands SB1 to SB4 extend up to the border frequency which is, for example, fCELP. Thus, all the subbands below the border frequency fCELP constitute the lower band and the frequency content above the border frequency constitutes the higher band.

    [0053] Particularly, the LPC analyzer 808 of Fig. 8 typically calculates shaping information for each subband individually. Thus, the LPC analyzer 808 preferably calculates four different kinds of subband information for the four subbands SB1 to SB4 so that each subband has its associated shaping information.

    [0054] Furthermore, the shaping is applied by the shaper 804 for each subband SB1 to SB4 using the shaping information calculated for exactly this subband and, importantly, a shaping for the higher band is also done, but the shaping information for the higher band is not being calculated due to the fact that the linear prediction analyzer calculating the shaping information receives a band limited signal band limited to the lower frequency band. Nevertheless, in order to also perform a shaping for the higher frequency band, the shaping information for subband SB4 is used for shaping the higher band. Thus, the shaper 804 is configured to weigh the spectral coefficients of the upper frequency band using a shaping factor calculated for a highest subband of the lower frequency band. The highest subband corresponding to SB4 in Fig. 9 has a highest center frequency among all center frequencies of subbands of the lower frequency band.

    [0055] Fig. 11 illustrates a preferred flowchart for explaining the functionality of the detector 802. Particularly, the detector 802 is configured to determine a peak spectral region in the upper frequency band, when at least one of a group of conditions is true, where the group of conditions comprises a low-band amplitude condition 1102, a peak distance condition 1104 and a peak amplitude condition 1106.

    [0056] Preferably, the different conditions are applied in exactly the order illustrated in Fig. 11. In other words, the low-band amplitude condition 1102 is calculated before the peak distance condition 1104, and the peak distance condition is calculated before the peak amplitude condition 1106. In a situation, where all three conditions must be true in order to detect the peak spectral region, a computationally efficient detector is obtained by applying the sequential processing in Fig. 11, where, as soon as a certain condition is not true, i.e., is false, the detection process for a certain time frame is stopped and it is determined that an attenuation of a peak spectral region in this time frame is not required. Thus, when it is already determined for a certain time frame that the low-band amplitude condition 1102 is not fulfilled, i.e., is false, then the control proceeds to the decision that an attenuation of a peak spectral region in this time frame is not necessary and the procedure goes on without any additional attenuation. When, however, the controller determines for condition 1102 that same is true, the second condition 1104 is determined. This peak distance condition is once again determined before the peak amplitude 1106 so that the control determines that no attenuation of the peak spectral region is performed, when condition 1104 results in a false result. Only when the peak distance condition 1104 has a true result, the third peak amplitude condition 1106 is determined.

    [0057] In other embodiments, more or less conditions can be determined, and a sequential or parallel determination can be performed, although the sequential determination as exemplarily illustrated in Fig. 11 is preferred in order to save computational resources that are particularly valuable in mobile applications that are battery powered.

    [0058] Figs. 12, 13, 14 provide preferred embodiments for the conditions 1102, 1104 and 1106.

    [0059] In the low-band amplitude condition, a maximum spectral amplitude in the lower band is determined as illustrated at block 1202. This value is max_low. Furthermore, in block 1204, a maximum spectral amplitude in the upper band is determined that is indicated as max_high.

    [0060] In block 1206, the determined values from blocks 1232 and 1234 are processed preferably together with a predetermined number c1 in order to obtain the false or true result of condition 1102. Preferably, the conditions in blocks 1202 and 1204 are performed before shaping with the lower band shaping information, i.e., before the procedure performed by the spectral shaper 804 or, with respect to Fig. 10, 804a.

    [0061] With respect to the predetermined number c1 of Fig. 12 used in block 1206, a value of 16 is preferred, but values between 4 and 30 have been proven useful as well.
    Fig. 13 illustrates a preferred embodiment of the peak distance condition. In block 1302, a first maximum spectral amplitude in the lower band is determined that is indicated as max_low.

    [0062] Furthermore, a first spectral distance is determined as illustrated at block 1304. This first spectral distance is indicated as dist_low. Particularly, the first spectral distance is a distance of the first maximum spectral amplitude as determined by block 1302 from a border frequency between a center frequency of the lower frequency band and a center frequency of the upper frequency band. Preferably, the border frequency is f_celp, but this frequency can have any other value as outlined before.

    [0063] Furthermore, block 1306 determines a second maximum spectral amplitude in the upper band that is called max_high. Furthermore, a second spectral distance 1308 is determined and indicated as dist_high. The second spectral distance of the second maximum spectral amplitude from the border frequency is once again preferably determined with spectral f_celp as the border frequency.

    [0064] Furthermore, in block 1310, it is determined whether the peak distance condition is true, when the first maximum spectral amplitude weighted by the first spectral distance and weighted by a predetermined number being greater than 1 is greater than the second maximum spectral amplitude weighted by the second spectral distance.

    [0065] Preferably, a predetermined number c2 is equal to 4 in the most preferred embodiment. Values between 1.5 and 8 have been proven as useful.

    [0066] Preferably, the determination in block 1302 and 1306 is performed after shaping with the lower band shaping information, i.e., subsequent to block 804a, but, of course, before block 804b in Fig. 10.

    [0067] Fig. 14 illustrates a preferred implementation of the peak amplitude condition. Particularly, block 1402 determines a first maximum spectral amplitude in the lower band and block 1404 determines a second maximum spectral amplitude in the upper band where the result of block 1402 is indicated as max_low2 and the result of block 1404 is indicated as max_high.

    [0068] Then, as illustrated in block 1406, the peak amplitude condition is true, when the second maximum spectral amplitude is greater than the first maximum spectral amplitude weighted by a predetermined number c3 being greater than or equal to 1. c3 is preferably set to a value of 1.5 or to a value of 3 depending on different rates where, generally, values between 1.0 and 5.0 have been proven as useful.

    [0069] Furthermore, as indicated in Fig. 14, the determination in blocks 1402 and 1404 takes place after shaping with the low-band shaping information, i.e., subsequent to the processing illustrated in block 804a and before the processing illustrated by block 804b or, with respect to Fig. 17, subsequent to block 1702 and before block 1704.

    [0070] In other embodiments, the peak amplitude condition 1106 and, particularly, the procedure in Fig. 14, block 1402 is not determined from the smallest value in the lower frequency band, i.e., the lowest frequency value of the spectrum, but the determination of the first maximum spectral amplitude in the lower band is determined based on a portion of the lower band where the portion extends from a predetermined start frequency until a maximum frequency of the lower frequency band, where the predetermined start frequency is greater than a minimum frequency of the lower frequency band. In an embodiment, the predetermined start frequency is at least 10% of the lower frequency band above the minimum frequency of the lower frequency band or, in other embodiments, the predetermined start frequency is at a frequency being equal to half a maximum frequency of the lower frequency band within a tolerance range of plus or minus 10% of half the maximum frequency.

    [0071] Furthermore, it is preferred that the third predetermined number c3 depends on a bitrate to be provided by the quantizer/coder stage, so that the predetermined number is higher for a higher bitrate. In other words, when the bitrate that has to be provided by the quantizer and coder stage 806 is high, then c3 is high, while, when the bitrate is to be determined as low, then the predetermined number c3 is low. When the preferred equation in block 1406 is considered, it becomes clear that the higher predetermined number c3 is, the peak spectral region is determined more rarely. When, however, c3 is small, then a peak spectral region where there are spectral values to be finally attenuated is determined more often.

    [0072] Blocks 1202, 1204, 1402, 1404 or 1302 and 1306 always determine a spectral amplitude. The determination of the spectral amplitude can be performed differently. One way of the determination of the spectral envelope is the determination of an absolute value of a spectral value of the real spectrum. Alternatively, the spectral amplitude can be a magnitude of a complex spectral value. In other embodiments, the spectral amplitude can be any power of the spectral value of the real spectrum or any power of a magnitude of a complex spectrum, where the power is greater than 1. Preferably, the power is an integer number, but powers of 1.5 or 2.5 additionally have proven to be useful. Preferably, nevertheless, powers of 2 or 3 are preferred.

    [0073] Generally, the shaper 804 is configured to attenuate at least one spectral value in the detected peak spectral region based on a maximum spectral amplitude in the upper frequency band and/or based on a maximum spectral amplitude in the lower frequency band. In other embodiments, the shaper is configured to determine the maximum spectral amplitude in a portion of the lower frequency band, the portion extending from a predetermined start frequency of the lower frequency band until a maximum frequency of the lower frequency band. The predetermined start frequency is greater than a minimum frequency of the lower frequency band and is preferably at least 10% of the lower frequency band above the minimum frequency of the lower frequency band or the predetermined start frequency is preferably at the frequency being equal to half of a maximum frequency of the lower frequency band within a tolerance of plus or minus 10% of half of the maximum frequency.

    [0074] The shaper furthermore is configured to determine the attenuation factor determining the additional attenuation, where the attenuation factor is derived from the maximum spectral amplitude in the lower frequency band multiplied by a predetermined number being greater than or equal to one and divided by the maximum spectral amplitude in the upper frequency band. To this end, reference is made to block 1602 illustrating the determination of a maximum spectral amplitude in the lower band (preferably after shaping, i.e., after block 804a in Fig. 10 or after block 1702 in Fig. 17).

    [0075] Furthermore, the shaper is configured to determine the maximum spectral amplitude in the higher band, again preferably after shaping as, for example, is done by block 804a in Fig. 10 or block 1702 in Fig. 17. Then, in block 1606, the attenuation factor fac is calculated as illustrated, where the predetermined number c3 is set to be greater than or equal to 1. In embodiments, c3 in Fig. 16 is the same predetermined number c3 as in Fig. 14. However, in other embodiments, c3 in Fig. 16 can be set different from c3 in Fig. 14. Additionally, c3 in Fig. 16 that directly influences the attenuation factor is also dependent on the bitrate so that a higher predetermined number c3 is set for a higher bitrate to be done by the quantizer/coder stage 806 as illustrated in Fig. 8.

    [0076] Fig. 17 illustrates a preferred implementation similar to what is shown at Fig. 10 at blocks 804a and 804b, i.e., that a shaping with the low-band gain information applied to the spectral values above the border frequency such as fcelp is performed in order to obtain shaped spectral values above the border frequency and additionally in a following step 1704, the attenuation factor fac as calculated by block 1606 in Fig. 16 is applied in block 1704 of Fig. 17. Thus, Fig. 17 and Fig. 10 illustrate a situation where the shaper is configured to shape the spectral values in the detected spectral region based on a first weighting operation using a portion of the shaping information for the lower frequency band and a second subsequent weighting operation using an attenuation information, i.e., the exemplary attenuation factor fac.

    [0077] In other embodiments, however, the order of steps in Fig. 17 is reversed so that the first weighting operation takes place using the attenuation information and the second subsequent weighting information takes place using at least a portion of the shaping information for the lower frequency band. Or, alternatively, the shaping is performed using a single weighting operation using a combined weighting information depending and being derived from the attenuation information on the one hand and at least a portion of the shaping information for the lower frequency band on the other hand.

    [0078] As illustrated in Fig. 17, the additional attenuation information is applied to all the spectral values in the detected peak spectral region. Alternatively, the attenuation factor is only applied to, for example, the highest spectral value or the group of highest spectral values, where the members of the group can range from 2 to 10, for example. Furthermore, embodiments also apply the attenuation factor to all spectral values in the upper frequency band for which the peak spectral region has been detected by the detector for a time frame of the audio signal. Thus, in this embodiment, the same attenuation factor is applied to the whole upper frequency band when only a single spectral value has been determined as a peak spectral region.

    [0079] When, for a certain frame, no peak spectral region has been detected, then the lower frequency band and the upper frequency band are shaped by the shaper without any additional attenuation. Thus, a switching over from time frame to time frame is performed, where, depending on the implementation, some kind of smoothing of the attenuation information is preferred.

    [0080] Preferably, the quantizer and encoder stage comprise a rate loop processor as illustrated in Fig. 15a and Fig. 15b. In an embodiment, the quantizer and coder stage 806 comprises a global gain weighter 1502, a quantizer 1504 and an entropy coder such as an arithmetic or Huffman coder 1506. Furthermore, the entropy coder 1506 provides, for a certain set of quantized values for a time frame, an estimated or measured bitrate to a controller 1508.

    [0081] The controller 1508 is configured to receive a loop termination criterion on the one hand and/or a predetermined bitrate information on the other hand. As soon as the controller 1508 determines that a predetermined bitrate is not obtained and/or a termination criterion is not fulfilled, then the controller provides an adjusted global gain to the global gain weighter 1502. Then, the global gain weighter applies the adjusted global gain to the shaped and attenuated spectral lines of a time frame. The global gain weighted output of block 1502 is provided to the quantizer 1504 and the quantized result is provided to the entropy encoder 1506 that once again determines an estimated or measured bitrate for the data weighted with the adjusted global gain. In case the termination criterion is fulfilled and/or the predetermined bitrate is fulfilled, then the encoded audio signal is output at output line 814. When, however, the predetermined bitrate is not obtained or a termination criterion is not fulfilled, then the loop starts again. This is illustrated in more detail in Fig. 15b.

    [0082] When the controller 1508 determines that the bitrate is too high as illustrated in block 1510, then a global gain is increased as illustrated in block 1512. Thus, all shaped and attenuated spectral lines become smaller since they are divided by the increased global gain and the quantizer then quantizes the smaller spectral values so that the entropy coder results in a smaller number of required bits for this time frame. Thus, the procedures of weighting, quantizing, and encoding is performed with the adjusted global gain as illustrated in block 1514 in Fig. 15b, and, then, once again it is determined whether the bitrate is too high. If the bitrate is still too high, then once again blocks 1512 and 1514 are performed. When, however, it is determined that the bitrate is not too high, the control proceeds to step 1516 that outlines, whether a termination criterion is fulfilled. When the termination criterion is fulfilled, the rate loop is stopped and the final global gain is additionally introduced into the encoded signal via an output interface such as the output interface 1014 of Fig. 10.

    [0083] When, however, it is determined that the termination criterion is not fulfilled, then the global gain is decreased as illustrated in block 1518 so that, in the end, the maximum bitrate allowed is used. This makes sure that time frames that are easy to encode are encoded with a higher precision, i.e., with less loss. Therefore, for such instances, the global gain is decreased as illustrated in block 1518 and step 1514 is performed with the decreased global gain and step 1510 is performed in order to look whether the resulting bitrate is too high or not.

    [0084] Naturally, the specific implementation regarding the global gain increase or decrease increment can be set as required. Additionally, the controller 1508 can be implemented to either have blocks 1510, 1512 and 1514 or to have blocks 1510, 1516, 1518 and 1514. Thus, depending on the implementation, and also depending on the starting value for the global gain, the procedure can be such that, from a very high global gain it is started until the lowest global gain that still fulfills the bitrate requirements is found. On the other hand, the procedure can be done in such a way in that it is started from a quite low global gain and the global gain is increased until an allowable bitrate is obtained. Additionally, as illustrated in Fig. 15b, even a mix between both procedures can be applied as well.

    [0085] Fig. 10 illustrates the embedding of the inventive audio encoder consisting of blocks 802, 804a, 804b and 806 within a switched time domain/frequency domain encoder setting.

    [0086] Particularly, the audio encoder comprises a common processor. The common processor consists of an ACELP/TCX controller 1004 and the band limiter such as a resampler 1006 and an LPC analyzer 808. This is illustrated by the hatched boxes indicated by 1002.

    [0087] Furthermore, the band limiter feeds the LPC analyzer that has already been discussed with respect to Fig. 8. Then, the LPC shaping information generated by the LPC analyzer 808 is forwarded to a CELP coder 1008 and the output of the CELP coder 1008 is input into an output interface 1014 that generates the finally encoded signal 1020. Furthermore, the time domain coding branch consisting of coder 1008 additionally comprises a time domain bandwidth extension coder 1010 that provides information and, typically, parametric information such as spectral envelope information for at least the high band of the full band audio signal input at input 1001. Preferably, the high band processed by the time domain band width extension coder 1010 is a band starting at the border frequency that is also used by the band limiter 1006. Thus, the band limiter performs a low pass filtering in order to obtain the lower band and the high band filtered out by the low pass band limiter 1006 is processed by the time domain band width extension coder 1010.

    [0088] On the other hand, the spectral domain or TCX coding branch comprises a time-spectrum converter 1012 and exemplarily, a tonal mask as discussed before in order to obtain a gap-filling encoder processing.

    [0089] Then, the result of the time-spectrum converter 1012 and the additional optional tonal mask processing is input into a spectral shaper 804a and the result of the spectral shaper 804a is input into an attenuator 804b. The attenuator 804b is controlled by the detector 802 that performs a detection either using the time domain data or using the output of the time-spectrum convertor block 1012 as illustrated at 1022. Blocks 804a and 804b together implement the shaper 804 of Fig. 8 as has been discussed previously. The result of block 804 is input into the quantizer and coder stage 806 that is, in a certain embodiment, controlled by a predetermined bitrate. Additionally, when the predetermined numbers applied by the detector also depend on the predetermined bitrate, then the predetermined bitrate is also input into the detector 802 (not shown in Fig. 10).

    [0090] Thus, the encoded signal 1020 receives data from the quantizer and coder stage, control information from the controller 1004, information from the CELP coder 1008 and information from the time domain bandwidth extension coder 1010.

    [0091] Subsequently, preferred embodiments of the present invention are discussed in even more detail.

    [0092] An option, which saves interoperability and backward compatibility to existing implementations is to do an encoder-side pre-processing. The algorithm, as explained subsequently, analyzes the MDCT spectrum. In case significant signal components below fCELP are present and high peaks above fCELP are found, which potentially destroy the coding of the complete spectrum in the rate loop, these peaks above fCELP are attenuated. Although the attenuation can not be reverted on decoder-side, the resulting decoded signal is perceptually significantly more pleasant than before, where huge parts of the spectrum were zeroed out completely.

    [0093] The attenuation reduces the focus of the rate loop on the peaks above fCELP and allows that significant low-frequency MDCT coefficients survive the rate loop.

    [0094] The following algorithm describes the encoder-side pre-processing:
    1. 1) Detection of low-band content (e.g. 1102):
      The detection of low-band content analyzes, whether significant low-band signal portions are present. For this, the maximum amplitude of the MDCT spectrum below and above fCELP are searched on the MDCT spectrum before the application of inverse LPC shape gains. The search procedure returns the following values:
      1. a) max_low_pre: The maximum MDCT coefficient below fCELP, evaluated on the spectrum of absolute values before the application of inverse LPC shaping gains
      2. b) max_high_pre: The maximum MDCT coefficient above fCELP, evaluated on the spectrum of absolute values before the application of inverse LPC shaping gains For the decision, the following condition is evaluated:
        Condition 1: c1 * max_low_pre > max_high_pre If Condition 1 is true, a significant amount of low-band content is assumed, and the pre-processing is continued; If Condition 1 is false, the pre-processing is aborted. This makes sure that no damage is applied to high-band only signals, e.g. a sine-sweep when above fCELP.
        Pseudo-code:

        where

        XM is the MDCT spectrum before application of the inverse LPC gain shaping,

        LTCX(CELP) is the number of MDCT coefficients up to fCELP

        LTCX(BW) is the number of MDCT coefficients for the full MDCT spectrum


      In an example implementation c1 is set to 16, and fabs returns the absolute value.
    2. 2) Evaluation of peak-distance metric (e.g. 1104):
      A peak-distance metric analyzes the impact of spectral peaks above fCELP on the arithmetic coder. Thus, the maximum amplitude of the MDCT spectrum below and above fCELP are searched on the MDCT spectrum after the application of inverse LPC shaping gains, i.e. in the domain where also the arithmetic coder is applied. In addition to the maximum amplitude, also the distance from fCELP is evaluated. The search procedure returns the following values:
      1. a) max_low: The maximum MDCT coefficient below fCELP, evaluated on the spectrum of absolute values after the application of inverse LPC shaping gains
      2. b) dist_low: The distance of max_low from fCELP
      3. c) max_high: The maximum MDCT coefficient above fCELP, evaluated on the spectrum of absolute values after the application of inverse LPC shaping gains
      4. d) dist_high: The distance of max_high from fCELP
        For the decision, the following condition is evaluated:
        Condition 2: c2 * dist_high * max_high > dist_low * max_low
        If Condition 2 is true, a significant stress for the arithmetic coder is assumed, due to either a very high spectral peak or a high frequency of this peak. The high peak will dominate the coding-process in the Rate loop, the high frequency will penalize the arithmetic coder, since the arithmetic coder always runs from low to high frequencies, i.e. higher frequencies are inefficient to code. If Condition 2 is true, the pre-processing is continued. If Condition 2 is false, the pre-processing is aborted.

        where

        M is the MDCT spectrum after application of the inverse LPC gain shaping,

        LTCX(CELP) is the number of MDCT coefficients up to fCELP

        LTCX(BW) is the number of MDCT coefficients for the full MDCT spectrum


      In an example implementation c2 is set to 4.
    3. 3) Comparison of peak-amplitude (e.g. 1106):
      Finally, the peak-amplitudes in psycho-acoustically similar spectral regions are compared. Thus, the maximum amplitude of the MDCT spectrum below and above fCELP are searched on the MDCT spectrum after the application of inverse LPC shaping gains. The maximum amplitude of the MDCT spectrum below fCELP is not searched for the full spectrum, but only starting at flow > 0 Hz. This is to discard the lowest frequencies, which are psycho-acoustically most important and usually have the highest amplitude after the application of inverse LPC shaping gains, and to only compare components with a similar psycho-acoustical importance. The search procedure returns the following values:
      1. a) max_low2: The maximum MDCT coefficient below fCELP, evaluated on the spectrum of absolute values after the application of inverse LPC shaping gains starting from flow
      2. b) max_high: The maximum MDCT coefficient above fCELP, evaluated on the spectrum of absolute values after the application of inverse LPC shaping gains
        For the decision, the following condition is evaluated:
        Condition 3: max_high > c3 * max_low2
        If condition 3 is true, spectral coefficients above fCELP are assumed, which have significantly higher amplitudes than just below fCELP, and which are assumed costly to encode. The constant c3 defines a maximum gain, which is a tuning parameter. If Condition 2 is true, the pre-processing is continued. If Condition 2 is false, the pre-processing is aborted.
        Pseudo-code:



        where

        Llow is a offset corresponding to flow

        XM is the MDCT spectrum after application of the inverse LPC gain shaping,

        LTCX(CELP) is the number of MDCT coefficients up to fCELP

        LTCX(BW) is the number of MDCT coefficients for the full MDCT spectrum


      In an example implementation flow is set to LTCX(CELP)/2. In an example implementation c3 is set to 1.5 for low bitrates and set to 3.0 for high bitrates.
    4. 4) Attenuation of high peaks above fCELP (e.g. Figs. 16 and 17):
      If condition 1-3 are found to be true, an attenuation of the peaks above fCELP is applied. The attenuation allows a maximum gain c3 compared to a psycho-acoustically similar spectral region. The attenuation factor is calculated as follows:

      The attenuation factor is subsequently applied to all MDCT coefficients above fCELP.
    5. 5) Pseudo-code:

      where

      XM is the MDCT spectrum after application of the inverse LPC gain shaping,

      LTCX(CELP) is the number of MDCT coefficients up to fCELP

      LTCX(BW) is the number of MDCT coefficients for the full MDCT spectrum



    [0095] The encoder-side pre-processing significantly reduces the stress for the coding-loop while still maintaining relevant spectral coefficients above fCELP.

    [0096] Fig. 7 illustrates an MDCT spectrum of a critical frame after the application of inverse LPC shaping gains and above described encoder-side pre-processing. Dependent on the numerical values chosen for c1, c2 and c3 the resulting spectrum, which is subsequently fed into the rate loop, might look as above. They are significantly reduced, but still likely to survive the rate loop, without consuming all available bits.

    [0097] Although some aspects have been described in the context of an apparatus, it is clear that these aspects also represent a description of the corresponding method, where a block or device corresponds to a method step or a feature of a method step. Analogously, aspects described in the context of a method step also represent a description of a corresponding block or item or feature of a corresponding apparatus. Some or all of the method steps may be executed by (or using) a hardware apparatus, like for example, a microprocessor, a programmable computer or an electronic circuit. In some embodiments, one or more of the most important method steps may be executed by such an apparatus.

    [0098] The inventive encoded audio signal can be stored on a digital 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.

    [0099] Depending on certain implementation requirements, embodiments of the invention can be implemented in hardware or in software. The implementation can be performed using a non-transitory storage medium or a digital storage medium, for example a floppy disk, a DVD, a Blu-Ray, a CD, a ROM, a PROM, an EPROM, an EEPROM or a FLASH memory, having electronically readable control signals stored thereon, which cooperate (or are capable of cooperating) with a programmable computer system such that the respective method is performed. Therefore, the digital storage medium may be computer readable.

    [0100] Some embodiments according to the invention 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.

    [0101] Generally, embodiments of the present invention can be implemented as a computer program product with a program code, the program code being operative for performing one of the methods when the computer program product runs on a computer. The program code may for example be stored on a machine readable carrier.

    [0102] Other embodiments comprise the computer program for performing one of the methods described herein, stored on a machine readable carrier.

    [0103] In other words, an embodiment of the inventive method is, therefore, a computer program having a program code for performing one of the methods described herein, when the computer program runs on a computer.

    [0104] A further embodiment of the inventive methods is, therefore, a data carrier (or a digital storage medium, or a computer-readable medium) comprising, recorded thereon, the computer program for performing one of the methods described herein. The data carrier, the digital storage medium or the recorded medium are typically tangible and/or non-transitionary.

    [0105] A further embodiment of the inventive method is, therefore, a data stream or a sequence of signals representing the computer program for performing one of the methods described herein. The data stream or the sequence of signals may for example be configured to be transferred via a data communication connection, for example via the Internet.

    [0106] A further embodiment comprises a processing means, for example a computer, or a programmable logic device, configured to or adapted to perform one of the methods described herein.

    [0107] A further embodiment comprises a computer having installed thereon the computer program for performing one of the methods described herein.

    [0108] A further embodiment according to the invention comprises an apparatus or a system configured to transfer (for example, electronically or optically) a computer program for performing one of the methods described herein to a receiver. The receiver may, for example, be a computer, a mobile device, a memory device or the like. The apparatus or system may, for example, comprise a file server for transferring the computer program to the receiver.

    [0109] In some embodiments, a programmable logic device (for example a field programmable gate array) may be used to perform some or all of the functionalities of the methods described herein. In some embodiments, a field programmable gate array may cooperate with a microprocessor in order to perform one of the methods described herein. Generally, the methods are preferably performed by any hardware apparatus.

    [0110] The apparatus described herein may be implemented using a hardware apparatus, or using a computer, or using a combination of a hardware apparatus and a computer.

    [0111] The apparatus described herein, or any components of the apparatus described herein, may be implemented at least partially in hardware and/or in software.

    [0112] The methods described herein may be performed using a hardware apparatus, or using a computer, or using a combination of a hardware apparatus and a computer.

    [0113] The methods described herein, or any components of the apparatus described herein, may be performed at least partially by hardware and/or by software.

    [0114] The above described embodiments are merely illustrative for the principles of the present invention. It is understood that modifications and variations of the arrangements and the details described herein will be apparent to others skilled in the art. It is the intent, therefore, to be limited only by the scope of the impending patent claims and not by the specific details presented by way of description and explanation of the embodiments herein.

    [0115] In the foregoing description, it can be seen that various features are grouped together in embodiments for the purpose of streamlining the disclosure. This method of disclosure is not to be interpreted as reflecting an intention that the claimed embodiments require more features than are expressly recited in each claim. Rather, as the following claims reflect, inventive subject matter may lie in less than all features of a single disclosed embodiment. Thus the following claims are hereby incorporated into the Detailed Description, where each claim may stand on its own as a separate embodiment. While each claim may stand on its own as a separate embodiment, it is to be noted that - although a dependent claim may refer in the claims to a specific combination with one or more other claims - other embodiments may also include a combination of the dependent claim with the subject matter of each other dependent claim or a combination of each feature with other dependent or independent claims. Such combinations are proposed herein unless it is stated that a specific combination is not intended. Furthermore, it is intended to include also features of a claim to any other independent claim even if this claim is not directly made dependent to the independent claim.

    [0116] It is further to be noted that methods disclosed in the specification or in the claims may be implemented by a device having means for performing each of the respective steps of these methods.

    [0117] Furthermore, in some embodiments a single step may include or may be broken into multiple sub steps. Such sub steps may be included and part of the disclosure of this single step unless explicitly excluded.

    References



    [0118] 
    1. [1] 3GPP TS 26.445 - Codec for Enhanced Voice Services (EVS); Detailed algorithmic description

    Annex



    [0119] Subsequently, portions of the above standard release 13 (3GPP TS 26.445 - Codec for Enhanced Voice Services (EVS); Detailed algorithmic description) are indicated. Section 5.3..3.2.3 describes a preferred embodiment of the shaper, section 5.3.3.2.7 describes a preferred embodiment of the quantizer from the quantizer and coder stage, and section 5.3.3.2.8 describes an arithmetic coder in a preferred embodiment of the coder in the quantizer and coder stage, wherein the preferred rate loop for the constant bit rate and the global gain is described in section 5.3.2.8.1.2. The IGF features of the preferred embodiment are described in section 5.3.3.2.11, where specific reference is made to section 5.3.3.2.11.5.1 IGF tonal mask calculation. Other portions of the standard are incorporated by reference herein.

    5.3.3.2.3 LPC shaping in MDCT domain


    5.3.3.2.3.1 General Principle



    [0120] LPC shaping is performed in the MDCT domain by applying gain factors computed from weighted quantized LP filter coefficients to the MDCT spectrum. The input sampling rate srinp, on which the MDCT transform is based, can be higher than the CELP sampling rate srcelp, for which LP coefficients are computed. Therefore LPC shaping gains can only be computed for the part of the MDCT spectrum corresponding to the CELP frequency range. For the remaining part of the spectrum (if any) the shaping gain of the highest frequency band is used.

    5.3.3.2.3.2 Computation of LPC shaping gains



    [0121] To compute the 64 LPC shaping gains the weighted LP filter coefficients are first transformed into the frequency domain using an oddly stacked DFT of length 128:

    The LPC shaping gains gLPC are then computed as the reciprocal absolute values of XLPC :


    5.3.3.2.3.3 Applying LPC shaping gains to MDCT spectrum



    [0122] The MDCT coefficients XM corresponding to the CELP frequency range are grouped into 64 sub-bands. The coefficients of each sub-band are multiplied by the reciprocal of the corresponding LPC shaping gain to obtain the shaped spectrum M. If the number of MDCT bins corresponding to the CELP frequency range

    is not a multiple of 64, the width of sub-bands varies by one bin as defined by the following pseudo-code:



    [0123] The remaining MDCT coefficients above the CELP frequency range (if any) are multiplied by the reciprocal of the last LPC shaping gain:


    5.3.3.2.4 Adaptive low frequency emphasis


    5.3.3.2.4.1 General Principle



    [0124] The purpose of the adaptive low-frequency emphasis and de-emphasis (ALFE) processes is to improve the subjective performance of the frequency-domain TCX codec at low frequencies. To this end, the low-frequency MDCT spectral lines are amplified prior to quantization in the encoder, thereby increasing their quantization SNR, and this boosting is undone prior to the inverse MDCT process in the internal and external decoders to prevent amplification artifacts.

    [0125] There are two different ALFE algorithms which are selected consistently in encoder and decoder based on the choice of arithmetic coding algorithm and bit-rate. ALFE algorithm 1 is used at 9.6 kbps (envelope based arithmetic coder) and at 48 kbps and above (context based arithmetic coder). ALFE algorithm 2 is used from 13.2 up to incl. 32 kbps. In the encoder, the ALFE operates on the spectral lines in vector x [ ] directly before (algorithm 1) or after (algorithm 2) every MDCT quantization, which runs multiple times inside a rate-loop in case of the context based arithmetic coder (see subclause 5.3.3.2.8.1).

    5.3.3.2.4.2 Adaptive emphasis algorithm 1



    [0126] ALFE algorithm 1 operates based on the LPC frequency-band gains, lpcGains [ ]. First, the minimum and maximum of the first nine gains - the low-frequency (LF) gains - are found using comparison operations executed within a loop over the gain indices 0 to 8.

    [0127] Then, if the ratio between the minimum and maximum exceeds a threshold of 1/32, a gradual boosting of the lowest lines in x is performed such that the first line (DC) is amplified by (32 min/max)0,25 and the 33rd line is not amplified: tmp = 32 * min if ((max < tmp) && (max > 0)) { fac = tmp = pow(tmp / max, 1/128) for (i = 31; i >= 0; i--) { /* gradual boosting of lowest 32 lines */ x[i] *= fac fac *= tmp } }

    5.3.3.2.4.3 Adaptive emphasis algorithm 2



    [0128] ALFE algorithm 2, unlike algorithm 1, does not operate based on transmitted LPC gains but is signaled by means of modifications to the quantized low-frequency (LF) MDCT lines. The procedure is divided into five consecutive steps:
    • Step 1: first find first magnitude maximum at index i_max in lower spectral quarter

      / 4) utilizing invGain = 2/gTCX and modifying the maximum: xq[i_max] += (xq[i_max] < 0) ? -2 : 2
    • Step 2: then compress value range of all x [i] up to i_max by requantizing all lines at k = 0 ... i_max-1 as in the subclause describing the quantization, but utilizing invGain instead of gTCX as the global gain factor.
    • Step 3: find first magnitude maximum below i_max

      which is half as high if i_max > -1 using invGain = 4/gTCX and modifying the maximum: xq [i_max] += (xq [i_max] < 0) ? -2 : 2
    • Step 4: re-compress and quantize all x [i] up to the half-height i_max found in the previous step, as in step 2
    • Step 5: finish and always compress two lines at the latest i_max found, i.e. at k = i_max+1, i_max+2, again utilizing invGain = 2/gTCX if the initial i_max found in step 1 is greater than -1, or using invGain = 4/gTCX otherwise. All i_max are initialized to -1. For details please see AdaptLowFreqEmph () in tcx_utils_enc.c.

    5.3.3.2.5 Spectrum noise measure in power spectrum



    [0129] For guidance of quantization in the TXC encoding process, a noise measure between 0 (tonal) and 1 (noise-like) is determined for each MDCT spectral line above a specified frequency based on the current transform's power spectrum. The power spectrum XP(k) is computed from the MDCT coefficients XM(k) and the MDST XS(k) coefficients on the same time-domain signal segment and with the same windowing operation:



    [0130] Each noise measure in noiseFlags(k) is then calculated as follows. First, if the transform length changed (e.g. after a TCX transition transform following an ACELP frame) or if the previous frame did not use TCX20 coding (e.g. in case a shorter transform length was used in the last frame), all noiseFlags(k) up to

    are reset to zero. The noise measure start line kstart is initialized according to the following table 1.
    Table 1: Initialization table of kstart in noise measure
    Bitrate (kbps)9.613.216.424.4324836128
    bw=NB, WB 66 128 200 320 320 320 320 320
    bw=SWB,FB 44 96 160 320 320 256 640 640


    [0131] For ACELP to TCX transitions, kstart is scaled by 1.25. Then, if the noise measure start line kstart is less than

    the noiseFlags(k) at and above kstart are derived recursively from running sums of power spectral lines:





    [0132] Furthermore, every time noiseFlags(k) is given the value zero in the above loop, the variable lastTone is set to k. The upper 7 lines are treated separately since s(k) cannot be updated any more (c(k), however, is computed as above):



    [0133] The uppermost line at

    is defined as being noise-like, hence

    Finally, if the above variable lastTone (which was initialized to zero) is greater than zero, then noiseFlags(lastTone +1) = 0 . Note that this procedure is only carried out in TCX20, not in other TCX modes (noiseFlags(k) = 0 for

    )

    5.3.3.2.6 Low pass factor detector



    [0134] A low pass factor clpf is determined based on the power spectrum for all bitrates below 32.0 kbps. Therefore, the power spectrum XP(k) is compared iteratively against a threshold tlpf for all

    where tlpf = 32.0 for regular MDCT windows and tlpf = 64.0 for ACELP to MDCT transition windows. The iteration stops as soon as XP(k)>t/pf.

    [0135] The low pass factor clpf determines as

    where clpf, prev is the last determined low pass factor. At encoder startup, clpf,prev is set to 1.0. The low pass factor clpf is used to determine the noise filling stop bin (see subclause 5.3.3.2.10.2).

    5.3.3.2.7 Uniform quantizer with adaptive dead-zone



    [0136] For uniform quantization of the MDCT spectrum M after or before ALFE (depending on the applied emphasis algorithm, see subclause 5.3.3.2.4.1), the coefficients are first divided by the global gain gTCX (see subclause 5.3.3.2.8.1.1), which controls the step-size of quantization. The results are then rounded toward zero with a rounding offset which is adapted for each coefficient based on the coefficient's magnitude (relative to gTCX) and tonality (as defined by noiseFlags(k) in subclause 5.3.3.2.5). For high-frequency spectral lines with low tonality and magnitude, a rounding offset of zero is used, whereas for all other spectral lines, an offset of 0.375 is employed. More specifically, the following algorithm is executed.

    [0137] Starting from the highest coded MDCT coefficient at index

    we set M(k) = 0 and decrement k by 1 as long as condition noiseFlags(k) > 0 and |M(k)|/gTCX < 1 evaluates to true. Then downward from the first line at index k'≥ 0 where this condition is not met (which is guaranteed since noiseFlags(0) = 0), rounding toward zero with a rounding offset of 0.375 and limiting of the resulting integer values to the range -32768 to 32767 is performed:

    with k = 0..k'. Finally, all quantized coefficients of M(k) at and above

    are set to zero.

    5.3.3.2.8 Arithmetic coder



    [0138] The quantized spectral coefficients are noiselessly coded by an entropy coding and more particularly by an arithmetic coding.

    [0139] The arithmetic coding uses 14 bits precision probabilities for computing its code. The alphabet probability distribution can be derived in different ways. At low rates, it is derived from the LPC envelope, while at high rates it is derived from the past context. In both cases, a harmonic model can be added for refining the probability model.

    [0140] The following pseudo-code describes the arithmetic encoding routine, which is used for coding any symbol associated with a probability model. The probability model is represented by a cumulative frequency table cum_freq[]. The derivation of the probability model is described in the following subclauses. /* global varibles */ low high bits_to_follow ar_encode(symbol, cum_freq[]) { if (ari_first_symbol()) { low = 0; high - 65535; bits_to_follow = 0; } range = high-low+1; if (symbol > 0) { high = low + ((range*cum_freq[symbol-1])>>14) - 1; } low += ((range*cum_freq[symbol-1])>>14) - 1; for (;;) { if (high < 32768) { write_bit(0); while (bits_to_follow) { write_bit(1); bits_to_follow--; } } else if (low >= 32768) { write_bit (1) while (bits_to_follow) { write_bit(0); bits_to_follow--; } low -= 32768; high -= 32768; } else if ((low >= 16384) && (high < 49152)) { bits_to_follow += 1; low -= 16384; high -= 16384; } else break; low += low; high +- high+1; } if (ari_last_symbol()) /* flush bits */ if (low < 16384) { write_bit(0); while (bits_to_follow > 0) { write_bit(1); bits_to_follow--; } } else { write_bit(1); while (bits_to_follow > 0) { write_bit(0); bits_to_follow-- ; } } } } The helper functions ari_first_symbol() and ari_last_symbol() detect the first symbol and the last symbol of the generated codeword respectively.

    5.3.3.2.8.1 Context based arithmetic codec


    5.3.3.2.8.1.1 Global gain estimator



    [0141] The estimation of the global gain gTCX for the TCX frame is performed in two iterative steps. The first estimate considers a SNR gain of 6dB per sample per bit from SQ. The second estimate refines the estimate by taking into account the entropy coding.

    [0142] The energy of each block of 4 coefficients is first computed:



    [0143] A bisection search is performed with a final resolution of 0.125dB:

    Initialization: Set fac = offset = 12.8 and target = 0.15(target_bits - L/16)

    Iteration: Do the following block of operations 10 times

    1- fac=fac/2

    2- offset = offset - fac

    2-

    where

    3- if(ener>target) then offset=offset+fac



    [0144] The first estimate of gain is then given by:


    5.3.3.2.8.1.2 Rate-loop for constant bit rate and global gain



    [0145] In order to set the best gain gTCX within the constraints of used_bitstarget_bits, convergence process of gTCX and used_bits is carried out by using following valuables and constants:

    WLb and WUb denote weights corresponding to the lower bound the upper bound,

    gLb and gUb denote gain corresponding to the lower bound the upper bound, and

    Lb_found and Ub_found denote flags indicating gLb and gUb is found, respectively.

    µ and η are variables with µ=max(1,2.3-0.0025target_bits) and η=1/µ.

    λ and v are constants, set as 10 and 0.96.



    [0146] After the initial estimate of bit consumption by arithmetic coding, stop is set 0 when target_bits is larger than used_bits, while stop is set as used_bits when used_bits is larger than target_bits.

    [0147] If stop is larger than 0, that means used_bits is larger than target_bits,

    [0148] gTCX needs to be modified to be larger than the previous one and Lb_found is set as TRUE, gLb is set as the previous gTCX . WLb is set as



    [0149] When Ub_found was set, that means used_bits was smaller than target_bits, gTCX is updated as an interpolated value between upper bound and lower bound. ,



    [0150] Otherwise, that means Ub_found is FALSE, gain is amplified as

    with larger amplification ratio when the ratio of used_bits(=stop) and target_bits is larger to accelerate to attain gUb.

    [0151] If stop equals to 0, that means used_bits is smaller than target_bits,
    gTCX should be smaller than the previous one and Ub_found is set as 1, Ub is set as the previous gTCX and wUb is set as

    If Lb_found has been already set, gain is calculated as

    otherwise, in order to accelerate to lower band gain gLb, gain is reduced as,

    with larger reduction rates of gain when the ratio of used_bits and target_bits is small.

    [0152] After above correction of gain, quantization is performed and estimation of used_bits by arithmetic coding is obtained. As a result, stop is set 0 when target_bits is larger than used_bits, and is set as used_bits when it is larger than target_bits. If the loop count is less than 4, either lower bound setting process or upper bound setting process is carried out at the next loop depending on the value stop . If the loop count is 4, the final gain gTCX and the quantized MDCT sequence XQMDCT(k) are obtained.

    5.3.3.2.8.1.3 Probability model derivation and coding



    [0153] The quantized spectral coefficients X are noiselessly encoded starting from the lowest-frequency coefficient and progressing to the highest-frequency coefficient. They are encoded by groups of two coefficients a and b gathering in a so-called 2-tuple {a,b}.

    [0154] Each 2-tuple {a,b} is split into three parts namely, MSB, LSB and the sign. The sign is coded independently from the magnitude using uniform probability distribution. The magnitude itself is further divided in two parts, the two most significant bits (MSBs) and the remaining least significant bitplanes (LSBs, if applicable). The 2-tuples for which the magnitude of the two spectral coefficients is lower or equal to 3 are coded directly by the MSB coding. Otherwise, an escape symbol is transmitted first for signalling any additional bit plane.

    [0155] The relation between 2-tuple, the individual spectral values a and b of a 2-tuple, the most significant bit planes m and the remaining least significant bit planes, r, are illustrated in the example in figure 1. In this example three escape symbols are sent prior to the actual value m, indicating three transmitted least significant bit planes



    [0156] The probability model is derived from the past context. The past context is translated on a 12 bits-wise index and maps with the lookup table ari_context_lookup [] to one of the 64 available probability models stored in ari_cf_m[].

    [0157] The past context is derived from two 2-tuples already coded within the same frame. The context can be derived from the direct neighbourhood or located further in the past frequencies. Separate contexts are maintained for the peak regions (coefficients belonging to the harmonic peaks) and other (non-peak) regions according to the harmonic model. If no harmonic model is used, only the other (non-peak) region context is used.

    [0158] The zeroed spectral values lying in the tail of spectrum are not transmitted. It is achieved by transmitting the index of last non-zeroed 2-tuple. If harmonic model is used, the tail of the spectrum is defined as the tail of spectrum consisting of the peak region coefficients, followed by the other (non-peak) region coefficients, as this definition tends to increase the number of trailing zeros and thus improves coding efficiency. The number of samples to encode is computed as follows:



    [0159] The following data are written into the bitstream with the following order:
    1. 1- lastnz/2-1 is coded on

    2. 2- The entropy-coded MSBs along with escape symbols.
    3. 3- The signs with 1 bit-wise code-words
    4. 4- The residual quantization bits described in section when the bit budget is not fully used.
    5. 5- The LSBs are written backwardly from the end of the bitstream buffer.


    [0160] The following pseudo-code describes how the context is derived and how the bitstream data for the MSBs, signs and LSBs are computed. The input arguments are the quantized spectral coefficients X[], the size of the considered spectrum L, the bit budget target_bits, the harmonic model parameters (pi, hi), and the index of the last non zeroed symbol lastnz. ari_context_encode(X[], L, target_bits,pi[],hi[], lastnz) { c[0]=c[1]=p1=p2=0; for (k-0; k<lastnz; k+=2) { ari_copy_states(); (a1_i,pl, idx1) = get_next_coeff(pi,hi,lastnz); (b1_i,p2,idx2) = get_next_coeff(pi,hi,lastnz); t=get_context(idx1,idx2,c,p1,p2); esc_nb = levl = 0; a = al = abs(X[a1_i]); b = bl = abs (X[b1_i]); /* sign encoding*/ if (a1>0) save_bit(X[al_i]>0?0:1); if (b1>0) save_bit(X[b1_i]>0?0:1); /* MSB encoding */ while (al > 3 | | bl > 3) { pki = ari_context_lookup[t+1024*esc_nb]; /* write escape codeword */ ari_encode(17, ari_cf_m[pki]); a1>>=1; b1 >>=1; lev1++; esc_nb = min(levl,3); } pki = ari_context_lookup[t+1024*esc_nb]; ari_encode(a1+4*b1, ari_cf_m[pki]); /* LSB encoding */ for (lev=0; lev<lev1; lev++) { write bit_end((a>>lev)&1); write_bit_end((b>>lev)&1) ; } /*check budget*/ if (nbbits>target_bits) { ari_restore_states(); break; c=update_context(a,b,a1,b1,c,p1,p2); } write_sign_bits(); }

    [0161] The helper functions ari_save_states() and ari_restore_states() are used for saving and restoring the arithmetic coder states respectively. It allows cancelling the encoding of the last symbols if it violates the bit budget. Moreover and in case of bit budget overflow, it is able to fill the remaining bits with zeros till reaching the end of the bit budget or till processing lastnz samples in the spectrum.

    [0162] The other helper functions are described in the following subclauses.

    5.3.3.2.8.1.4 Get next coefficient



    [0163]  (a,p,idx) = get_next_coeff(pi, hi, lastnz) If ((ii[0] ≥ lastnz - min(#pi, lastnz)) or (ii[1] < min(#pi, lastnz) and pi[ii[1]] < hi[ii[0]])) then { p=1 idx=ii[1] a=pi[ii[1]] } else { p=0 idx=ii[0] + #pi a=hi[ii[0]] } ii[p]=ii[p] + 1

    [0164] The ii[0] and ii[1] counters are initialized to 0 at the beginning of ari_context_encode() (and ari_context_decode() in the decoder).

    5.3.3.2.8.1.5 Context update



    [0165] The context is updated as described by the following pseudo-code. It consists of the concatenation of two 4 bit-wise context elements.


    5.3.3.2.8.1.6 Get context



    [0166] The final context is amended in two ways:



    [0167] The context t is an index from 0 to 1023.

    5.3.3.2.8.1.7 Bit consumption estimation



    [0168] The bit consumption estimation of the context-based arithmetic coder is needed for the rate-loop optimization of the quantization. The estimation is done by computing the bit requirement without calling the arithmetic coder. The generated bits can be accurately estimated by: cum_freq= arith_cf_m[pki]+m proba*= cum_freq[0]- cum_freq[1] nlz=norm_l(proba) /*get the number of leading zero */ nbits=nlz proba»=14 where proba is an integer initialized to 16384 and m is a MSB symbol.

    5.3.3.2.8.1.8 Harmonic model



    [0169] For both context and envelope based arithmetic coding, a harmonic model is used for more efficient coding of frames with harmonic content. The model is disabled if any of the following conditions apply:
    • The bit-rate is not one of 9.6, 13.2, 16.4, 24.4, 32, 48 kbps.
    • The previous frame was coded by ACELP.
    • Envelope based arithmetic coding is used and the coder type is neither Voiced nor Generic.
    • The single-bit harmonic model flag in the bit-stream in set to zero.
    When the model is enabled, the frequency domain interval of harmonics is a key parameter and is commonly analysed and encoded for both flavours of arithmetic coders.

    5.3.3.2.8.1.8.1 Encoding of Interval of harmonics



    [0170] When pitch lag and gain are used for the post processing, the lag parameter is utilized for representing the interval of harmonics in the frequency domain. Otherwise, normal representation of interval is applied.

    5.3.3.2.8.1.8.1.1 Encoding interval depending on time domain pitch lag



    [0171] If integer part of pitch lag in time domain dint is less than the frame size of MDCT LTCX, frequency domain interval unit (between harmonic peaks corresponding to the pitch lag) TUNIT with 7 bit fractional accuracy is given by

    where dfr denotes the fractional part of pitch lag in time domain, res_max denotes the max number of allowable fractional values whose values are either 4 or 6 depending on the conditions.

    [0172] Since TUNIT has limited range, the actual interval between harmonic peaks in the frequency domain is coded relatively to TUNIT using the bits specified in table 2. Among candidate of multiplication factors, Ratio() given in the table 3 or table 4, the multiplication number is selected that gives the most suitable harmonic interval of MDCT domain transform coefficients.



    Table 2: Number of bits for specifying the multiplier depending on IndexT
    IndexT0123456789101112131415
    NB: 5 4 4 4 4 4 4 3 3 3 3 2 2 2 2 2
    WB: 5 5 5 5 5 5 4 4 4 4 4 4 4 2 2 2
    Table 3: Candidates of multiplier in the order of IndexMUL depending on IndexT (NB)
    IndexT                
    03456789101112131415161718
    19202122232425262728303234363840
    1 0.5 1 2 3 4 5 6 7 8 9 10 12 16 20 24 30
    2 2 3 4 5 6 7 8 9 10 12 14 16 18 20 24 30
    3 2 3 4 5 6 7 8 9 10 12 14 16 18 20 24 30
    4 2 3 4 5 6 7 8 9 10 12 14 16 18 20 24 30
    5 1 2 2.5 3 4 5 6 7 8 9 10 12 14 16 18 20
    6 1 1.5 2 2.5 3 3.5 4 4.5 5 6 7 8 9 10 12 16
    7 1 2 3 4 5 6 8 10 - - - - - - - -
    8 1 2 3 4 5 6 8 10 - - - - - - - -
    9 1 1.5 2 3 4 5 6 8 - - - - - - - -
    10 1 2 2.5 3 4 5 6 8 - - - - - - - -
    11 1 2 3 4 - - - - - - - - - - - -
    12 1 2 4 6 - - - - - - - - - - - -
    13 1 2 3 4 - - - - - - - - - - - -
    14 1 1.5 2 4 - - - - - - - - - - - -
    15 1 1.5 2 3 - - - - - - - - - - - -
    16 0.5 1 2 3 - - - - - - - - - - - -
    Table 4: Candidates of multiplier in the order of depending on IndexT (WB)
    IndexT                
    03456789101112131415161718
    19202122232425262728303234363840
    1 1 2 3 4 5 6 7 8 9 10 12 14 16 18 20 22
    24 26 28 30 32 34 36 38 40 44 48 54 60 68 78 80
    2 1.5 2 2.5 3 4 5 6 7 8 9 10 12 14 16 18 20
    22 24 26 28 30 32 34 36 38 40 42 44 48 52 54 68
    3 1 1.5 2 2.5 3 4 5 6 7 8 9 10 11 12 13 14
    15 16 18 20 22 24 26 28 30 32 34 36 40 44 48 54
    4 1 1.5 2 2.5 3 3.5 4 4.5 5 5.5 6 6.5 7 7.5 8 9
    10 11 12 13 14 15 16 18 20 22 24 26 28 34 40 41
    5 1 1.5 2 2.5 3 3.5 4 4.5 5 6 7 8 9 10 11 12
    13 14 15 16 17 18 19 20 21 22. 5 24 25 27 28 30 35
    6 0.5 1 1.5 2 2.5 3 3.5 4 4.5 5 5.5 6 7 8 9 10
    7 1 2 2.5 3 4 5 6 7 8 9 10 12 15 16 18 27
    8 1 1.5 2 2.5 3 3.5 4 5 6 8 10 15 18 22 24 26
    9 1 1.5 2 2.5 3 3.5 4 5 6 8 10 12 13 14 18 21
    10 0.5 1 1.5 2 2.5 3 4 5 6 8 9 11 12 13. 5 16 20
    11 0.5 1 1.5 2 2.5 3 4 5 6 7 8 10 11 12 14 20
    12 0.5 1 1.5 2 2,5 3 4 4.5 6 7.5 9 10 12 14 15 18
    13 0.5 1 1.2 5 1.5 1.7 5 2 2.5 3 3.5 4 4.5 5 6 8 9 14
    14 0.5 1 2 4 - - - - - - - - - - - -
    15 1 1.5 2 4 -   - - - - - - - - - -
    16 1 2 3 4   - - - - - - - - - - -

    5.3.3.2.8.1.8.1.2 Encoding interval without depending on time domain pitch lag



    [0173] When pitch lag and gain in the time domain is not used or the pitch gain is less than or equals to 0.46, normal encoding of the interval with un-equal resolution is used.

    [0174] Unit interval of spectral peaks TUNIT is coded as

    and actual interval TMDCT is represented with fractional resolution of Res as



    [0175] Each paramter is shown in table 5, where "small size" means when frame size is smaller than 256 of the target bit rates is less than or equal to 150.
    Table 5: Un-equal resolution for coding of (0<= index < 256)
     Resbasebias
    index < 16 3 6 0
    16 ≤ index < 80 4 8 16
    80 ≤ index < 208 3 12 80
    "small size" or 208 ≤ index < 224 1 28 208
    224 ≤ index < 256 0 188 224

    5.3.3.2.8.1.8.2 Void


    5.3.3.2.8.1.8.3 Search for interval of harmonics



    [0176] In search of the best interval of harmonics, encoder tries to find the index which can maximize the weighted sum EPERlOD of the peak part of absolute MDCT coefficients. EABSM (k) denotes sum of 3 samples of absolute value of MDCT domain transform coefficients as



    where num_peak is the maximum number that └n·TMDCT┘ reaches the limit of samples in the frequency domain.

    [0177] In case interval does not rely on the pitch lag in time domain, hierarchical search is used to save computational cost. If the index of the interval is less than 80, periodicity is checked by a coarse step of 4. After getting the best interval, finer periodicity is searched around the best interval from -2 to +2. If index is equal to or larger than 80, periodicity is searched for each index.

    5.3.3.2.8.1.8.4 Decision of harmonic model



    [0178] At the initial estimation, number of used bits without harmonic model, used_bits, and one with harmonic model, used_bitshm is obtained and the indicator of consumed bits IdicatorB are defined as





    where Index_bitshm denotes the additional bits for modelling harmonic structure, and stop and stophm indicate the consumed bits when they are larger than the target bits. Thus, the larger IdicatorB, the more preferable to use harmonic model. Relative periodicity indicatorhm is defined as the normalized sum of absolute values for peak regions of the shaped MDCT coefficients as

    where TMDCT_max is the harmonic interval that attain the max value of EPERIOD. When the score of periodicity of this frame is larger than the threshold as

    this frame is considered to be coded by the harmonic model. The shaped MDCT coefficients divided by gain gTCX are quantized to produce a sequence of integer values of MDCT coefficients, TCX_hm, and compressed by arithmetic coding with harmonic model. This process needs iterative convergence process (rate loop) to get gTCX and TCX_hm with consumed bits Bhm. At the end of convergence, in order to validate harmonic model, the consumed bits Bno_hm by arithmetic coding with normal (non-harmonic) model for TCX_hm is additionally calculated and compared with Bhm. If Bhm is larger than Bno_hm, arithmetic coding of TCX_hm is revert to use normal model. Bhm-Bno_hm can be used for residual quantization for further enhancements. Otherwise, harmonic model is used in arithmetic coding.

    [0179] In contrast, if the indicator of periodicity of this frame is smaller than or the same as the threshold, quantization and arithmetic coding are carried out assuming the normal model to produce a sequence of integer values of the shaped MDCT coefficients, TCX_no_hm with consumed bits Bno_hm. After convergence of rate loop, consumed bits Bhm by arithmetic coding with harmonic model for TCX_no_hm is calculated. If Bno_hm is larger than Bhm , arithmetic coding of TCX_nohm is switched to use harmonic model. Otherwise, normal model is used in arithmetic coding.

    5.3.3.2.8.1.9 Use of harmonic information in Context based arithmetic coding



    [0180] For context based arithmetic coding, all regions are classified into two categories. One is peak part and consists of 3 consecutive samples centered at Uth (U is a positive integer up to the limit) peak of harmonic peak of τU,



    [0181] The other samples belong to normal or valley part. Harmonic peak part can be specified by the interval of harmonics and integer multiples of the interval. Arithmetic coding uses different contexts for peak and valley regions.

    [0182] For ease of description and implementation, the harmonic model uses the following index sequences:





    In case of disabled harmonic model, these sequences are pi = ( ), and hi = ip = (0,..., LM-1).

    5.3.3.2.8.2 Envelope based arithmetic coder



    [0183] In the MDCT domain, spectral lines are weighted with the perceptual model W(z) such that each line can be quantized with the same accuracy. The variance of individual spectral lines follow the shape of the linear predictor A-1(z) weighted by the perceptual model, whereby the weighted shape is S(z)=W(z)A-1(z). W(z) is calculated by transforming

    to frequency domain LPC gains as detailed in subclauses 5.3.3.2.4.1 and 5.3.3.2.4.2. A-1 (z) is derived from

    after conversion to direct-form coefficients, and applying tilt compensation 1-γz-1, and finally transforming to frequency domain LPC gains. All other frequency-shaping tools, as well as the contribution from the harmonic model, shall be also included in this envelope shape S(z). Observe that this gives only the relative variances of spectral lines, while the overall envelope has arbitrary scaling, whereby we must begin by scaling the envelope.

    5.3.3.2.8.2.1 Envelope scaling



    [0184] We will assume that spectral lines xk are zero-mean and distributed according to the Laplace-distribution, whereby the probability distribution function is



    [0185] The entropy and thus the bit-consumption of such a spectral line is bitsk = 1 + log2 2ebk . However, this formula assumes that the sign is encoded also for those spectral lines which are quantized to zero. To compensate for this discrepancy, we use instead the approximation

    which is accurate for bk ≥ 0.08. We will assume that the bit-consumption of lines with bk ≤ 0.08 is bitsk = log2(1.0224) which matches the bit-consumption at bk = 0.08 . For large bk > 255 we use the true entropy bitsk = log2(2ebk) for simplicity.

    [0186] The variance of spectral lines is then

    If

    is the k th element of the power of the envelope shape |S(z)|2 then

    describes the relative energy of spectral lines such that

    where γ is scaling coefficient. In other words,

    describes only the shape of the spectrum without any meaningful magnitude and γ is used to scale that shape to obtain the actual variance



    [0187] Our objective is that when we encode all lines of the spectrum with an arithmetic coder, then the bit-consumption matches a pre-defined level B, that is,

    We can then use a bi-section algorithm to determine the appropriate scaling factor γ such that the target bit-rate B is reached.

    [0188] Once the envelope shape bk has been scaled such that the expected bit-consumption of signals matching that shape yield the target bit-rate, we can proceed to quantizing the spectral lines.

    5.3.3.2.8.2.2 Quantization rate loop



    [0189] Assume that xk is quantized to an integer k such that the quantization interval is [k-0.5, k+0.5] then the probability of a spectral line occurring in that interval is for |k| ≥ 1

    and for |k| = 0



    [0190] It follows that the bit-consumption for these two cases is in the ideal case



    [0191] By pre-computing the terms

    and

    we can efficiently calculate the bit-consumption of the whole spectrum.

    [0192] The rate-loop can then be applied with a bi-section search, where we adjust the scaling of the spectral lines by a factor ρ, and calculate the bit-consumption of the spectrum ρxk, until we are sufficiently close to the desired bit-rate. Note that the above ideal-case values for the bit-consumption do not necessarily perfectly coincide with the final bit-consumption, since the arithmetic codec works with a finite-precision approximation. This rate-loop thus relies on an approximation of the bit-consumption, but with the benefit of a computationally efficient implementation.

    [0193] When the optimal scaling σ has been determined, the spectrum can be encoded with a standard arithmetic coder. A spectral line which is quantized to a value k ≠ 0 is encoded to the interval

    and k = 0 is encoded onto the interval



    [0194] The sign of xk ≠ 0 will be encoded with one further bit

    [0195] Observe that the arithmetic coder must operate with a fixed-point implementation such that the above intervals are bit-exact across all platforms. Therefore all inputs to the arithmetic coder, including the linear predictive model and the weighting filter, must be implemented in fixed-point throughout the system

    5.3.3.2.8.2.3 Probability model derivation and coding



    [0196] When the optimal scaling σ has been determined, the spectrum can be encoded with a standard arithmetic coder. A spectral line which is quantized to a value k ≠ 0 is encoded to the interval

    and k = 0 is encoded onto the interval



    [0197] The sign of xk ≠ 0 will be encoded with one further bit.

    5.3.3.2.8.2.4 Harmonic model in envelope based arithmetic coding



    [0198] In case of envelope base arithmetic coding, harmonic model can be used to enhance the arithmetic coding. The similar search procedure as in the context based arithmetic coding is used for estimating the interval between harmonics in the MDCT domain. However, the harmonic model is used in combination of the LPC envelope as shown in figure 2. The shape of the envelope is rendered according to the information of the harmonic analysis.

    [0199] Harmonic shape at k in the frequency data sample is defined as

    when τ-4≤kτ+4, otherwise Q(k) = 1.0, where τ denotes center position of Uth harmonics.

    h and σ are height and width of each harmonics depending on the unit interval as shown,





    [0200] Height and width get larger when interval gets larger.

    [0201] The spectral envelope S(k) is modified by the harmonic shape Q(k) at k as

    where gain for the harmonic components gharm is always set as 0.75 for Generic mode, and gharm is selected from {0.6, 1.4, 4.5, 10.0} that minimizes Enorm for Voiced mode using 2 bits,






    5.3.3.2.9 Global gain coding


    5.3.3.2.9.1 Optimizing global gain



    [0202] The optimum global gain gopt is computed from the quantized and unquantized MDCT coefficients. For bit rates up to 32 kbps, the adaptive low frequency de-emphasis (see subclause 6.2.2.3.2) is applied to the quantized MDCT coefficients before this step. In case the computation results in an optimum gain less than or equal to zero, the global gain gTCX determined before (by estimate and rate loop) is used.




    5.3.3.2.9.2 Quantization of global gain



    [0203] For transmission to the decoder the optimum global gain gopt is quantized to a 7 bit index ITCX,gain:



    [0204] The dequantized global gain TCX is obtained as defined in subclause 6.2.2.3.3).

    5.3.3.2.9.3 Residual coding



    [0205] The residual quantization is a refinement quantization layer refining the first SQ stage. It exploits eventual unused bits target_bits-nbbits, where nbbits is the number of bits consumed by the entropy coder. The residual quantization adopts a greedy strategy and no entropy coding in order to stop the coding whenever the bitstream reaches the desired size.

    [0206] The residual quantization can refine the first quantization by two means. The first mean is the refinement of the global gain quantization. The global gain refinement is only done for rates at and above 13.2kbps. At most three additional bits is allocated to it. The quantized gain TCX is refined sequentially starting from n=0 and incrementing n by one after each following iteration:



    [0207] The second mean of refinement consists of re-quantizing the quantized spectrum line per line. First, the non-zeroed quantized lines are processed with a 1 bit residual quantizer:



    [0208] Finally, if bits remain, the zeroed lines are considered and quantized with on 3 levels. The rounding offset of the SQ with deadzone was taken into account in the residual quantizer design:


    5.3.3.2.10 Noise Filling



    [0209] On the decoder side noise filling is applied to fill gaps in the MDCT spectrum where coefficients have been quantized to zero. Noise filling inserts pseudo-random noise into the gaps, starting at bin kNFstart up to bin kNFstop -1. To control the amount of noise inserted in the decoder, a noise factor is computed on encoder side and transmitted to the decoder.

    5.3.3.2.10.1 Noise Filling Tilt



    [0210] To compensate for LPC tilt, a tilt compensation factor is computed. For bitrates below 13.2 kbps the tilt compensation is computed from the direct form quantized LP coefficients , while for higher bitrates a constant value is used:




    5.3.3.2.10.2 Noise Filling Start and Stop Bins



    [0211] The noise filling start and stop bins are computed as follows:




    5.3.3.2.10.3 Noise Transition Width



    [0212] At each side of a noise filling segment a transition fadeout is applied to the inserted noise. The width of the transitions (number of bins) is defined as:

    where HM denotes that the harmonic model is used for the arithmetic codec and previous denotes the previous codec mode.

    5.3.3.2.10.4 Computation of Noise Segments



    [0213] The noise filling segments are determined, which are the segments of successive bins of the MDCT spectrum between kNFstart and kNFstop,LP for which all coefficients are quantized to zero. The segments are determined as defined by the following pseudo-code:

    where kNF0(j) and kNF1(j) are the start and stop bins of noise filling segment j, and nNF is the number of segments.

    5.3.3.2.10.5 Computation of Noise Factor



    [0214] The noise factor is computed from the unquantized MDCT coefficients of the bins for which noise filling is applied.

    [0215] If the noise transition width wNF is 3 or less bins, an attenuation factor is computed based on the energy of even and odd MDCT bins:







    [0216] For each segment an error value is computed from the unquantized MDCT coefficients, applying global gain, tilt compensation and transitions:



    [0217] A weight for each segment is computed based on the width of the segment:



    [0218] The noise factor is then computed as follows:


    5.3.3.2.10.6 Quantization of Noise Factor



    [0219] For transmission the noise factor is quantized to obtain a 3 bit index:


    5.3.3.2.11 Intelligent Gap Filling



    [0220] The Intelligent Gap Filling (IGF) tool is an enhanced noise filling technique to fill gaps (regions of zero values) in spectra. These gaps may occur due to coarse quantization in the encoding process where large portions of a given spectrum might be set to zero to meet bit constraints. However, with the IGF tool these missing signal portions are reconstructed on the receiver side (RX) with parametric information calculated on the transmission side (TX). IGF is used only if TCX mode is active.

    [0221] See table 6 below for all IGF operating points:
    Table 6: IGF application modes
    BitrateMode
    9.6 kbps WB
    9.6 kbps SWB
    13.2 kbps SWB
    16.4 kbps SWB
    24.4 kbps SWB
    32.2 kbps SWB
    48.0 kbps SWB
    16.4 kbps FB
    24.4 kbps FB
    32.0 kbps FB
    48.0 kbps FB
    96.0 kbps FB
    128.0 kbps FB


    [0222] On transmission side, IGF calculates levels on scale factor bands, using a complex or real valued TCX spectrum. Additionally spectral whitening indices are calculated using a spectral flatness measurement and a crest-factor. An arithmetic coder is used for noiseless coding and efficient transmission to receiver (RX) side.

    5.3.3.2.11.1 IGF helper functions


    5.3.3.2.11.1.1 Mapping values with the transition factor



    [0223] If there is a transition from CELP to TCX coding (isCelpToTCX = true) or a TCX 10 frame is signalled ( isTCX10 = true), the TCX frame length may change. In case of frame length change, all values which are related to the frame length are mapped with the function tF :

    where n is a natural number, for example a scale factor band offset, and f is a transition factor, see table 11.

    5.3.3.2.11.1.2 TCX power spectrum



    [0224] The power spectrum P ∈ Pn of the current TCX frame is calculated with:

    where n is the actual TCX window length, R ∈ Pn is the vector containing the real valued part (cos-transformed) of the current TCX spectrum, and I ∈ Pn is the vector containing the imaginary (sin-transformed) part of the current TCX spectrum.

    5.3.3.2.11.1.3 The spectral flatness measurement function SFM



    [0225] Let P ∈ Pn be the TCX power spectrum as calculated according to subclause 5.3.3.2.11.1.2 and b the start line and e the stop line of the SFM measurement range.

    [0226] The SFM function, applied with IGF, is defined with:

    where n is the actual TCX window length and p is defined with:


    5.3.3.2.11.1.4 The crest factor function CREST



    [0227] Let P ∈ Pn be the TCX power spectrum as calculated according to subclause 5.3.3.2.11.1.2 and b the start line and e the stop line of the crest factor measurement range.

    [0228] The CREST function, applied with IGF, is defined with:

    where n is the actual TCX window length and Emax is defined with:


    5.3.3.2.11.1.5 The mapping function hT



    [0229] The hT mapping function is defined with:

    where s is a calculated spectral flatness value and k is the noise band in scope. For threshold values ThMk , ThSk refer to table 7 below.
    Table 7: Thresholds for whitening for nT, ThM and ThS
    BitrateModenTThMThS
    9.6 kbps WB 2 0.36, 0.36 1.41, 1.41
    9.6 kbps SWB 3 0.84, 0.89, 0.89 1.30, 1.25, 1.25
    13.2 kbps SWB 2 0.84, 0.89 1.30, 1.25
    16.4 kbps SWB 3 0.83, 0.89, 0.89 1.31, 1.19, 119
    24.4 kbps SWB 3 0.81, 0.85, 0.85 1.35, 1.23, 1.23
    32.2 kbps SWB 3 0.91, 0.85, 0.85 1.34, 1.35, 1.35
    48.0 kbps SWB 1 1.15 1.19
    16.4 kbps FB 3 0.63, 0.27, 0.36 1.53, 1.32, 0.67
    24.4 kbps FB 4 0.78, 0.31, 0.34, 0.34 1.49, 1.38, 0.65, 0.65
    32.0 kbps FB 4 0.78, 0.31, 0.34, 0.34 1.49, 1.38, 0.65, 0.65
    48.0 kbps FB 1 0.80 1.0
    96.0 kbps FB 1 0 2.82
    128.0 kbps FB 1 0 2.82

    5.3.3.2.11.1.6 Void


    5.3.3.2.11.1.7 IGF scale factor tables



    [0230] IGF scale factor tables are available for all modes where IGF is applied.
    Table 8: Scale factor band offset table
    BitrateModeNumber of bands (nB)Scale factor band offsets (t[0],t[1],...,t[nB])
    9.6 kbps WB 3 164, 186, 242, 320
    9.6 kbps SWB 3 200, 322, 444, 566
    13.2 kbps SWB 6 256, 288, 328, 376, 432, 496, 566
    16.4 kbps SWB 7 256, 288, 328, 376, 432, 496, 576, 640
    24.4 kbps SWB 8 256, 284, 318, 358, 402, 450, 508, 576, 640
    32.2 kbps SWB 8 256, 284, 318, 358, 402, 450, 508, 576, 640
    48.0 kbps SWB 3 512, 534, 576, 640
    16.4 kbps FB 9 256, 288, 328, 376, 432, 496, 576, 640, 720, 800
    24.4 kbps FB 10 256, 284, 318, 358, 402, 450, 508, 576, 640, 720, 800
    32.0 kbps FB 10 256, 284, 318, 358, 402, 450, 508, 576, 640, 720, 800
    48.0 kbps FB 4 512, 584, 656, 728, 800
    96.0 kbps FB 2 640, 720, 800
    128.0 kbps FB 2 640, 720, 800


    [0231] The table 8 above refers to the TCX 20 window length and a transition factor 1.00.

    [0232] For all window lengths apply the following remapping

    where tF is the transition factor mapping function described in subclause 5.3.3.2.11.1.1.

    5.3.3.2.11.1.8 The mapping function m



    [0233] 
    Table 9: IGF minimal source subband, minSb
    BitratemodeminSb
    9.6 kbps WB 30
    9.6 kbps SWB 32
    13.2 kbps SWB 32
    16.4 kbps SWB 32
    24.4 kbps SWB 32
    32.2 kbps SWB 32
    48.0 kbps SWB 64
    16.4 kbps FB 32
    24.4 kbps FB 32
    32.0 kbps FB 32
    48.0 kbps FB 64
    96.0 kbps FB 64
    128.0 kbps FB 64


    [0234] For every mode a mapping function is defined in order to access source lines from a given target line in IGF range.
    Table 10: Mapping functions for every mode
    BitrateModenTmapping Function
    9.6 kbps WB 2 m2a
    9.6 kbps SWB 3 m3a
    13.2 1kbps SWB 2 m2b
    16.4 kbps SWB 3 m3b
    24.4 kbps SWB 3 m3c
    32.2 kbps SWB 3 m3c
    48.0 kbps SWB 1 m1
    16.4 kbps FB 3 m3d
    24.4 kbps FB 4 m4
    32.0 kbps FB 4 m4
    48.0 kbps FB 1 m1
    96.0 kbps FB 1 m1
    128.0 kbps FB 1 m1


    [0235] The mapping function m1 is defined with:



    [0236] The mapping function m2a is defined with:



    [0237] The mapping function m2b is defined with:



    [0238] The mapping function m3a is defined with:



    [0239] The mapping function m3b is defined with:



    [0240] The mapping function m3c is defined with:



    [0241] The mapping function m3d is defined with:



    [0242] The mapping function m4 is defined with:



    [0243] The value f is the appropriate transition factor, see table 11 and tF is described in subclause 5.3.3.2.11.1.1.

    [0244] Please note, that all values t(0),t(1),..,t(nB) shall be already mapped with the function tF, as described in subclause 5.3.3.2.11.1.1. Values for nB are defined in table 8.

    [0245] The here described mapping functions will be referenced in the text as "mapping function m" assuming, that the proper function for the current mode is selected.

    5.3.3.2.11.2 IGF input elements (TX)



    [0246] The IGF encoder module expects the following vectors and flags as an input:

    R : vector with real part of the current TCX spectrum XM

    I : vector with imaginary part of the current TCX spectrum XS

    P : vector with values of the TCX power spectrum XP

    isTransient : flag, signalling if the current frame contains a transient, see subclause 5.3.2.4.1.1

    isTCX10: flag, signalling a TCX 10 frame

    isTCX20 : flag, signalling a TCX 20 frame

    isCelpToTCX : flag, signalling CELP to TCX transition; generate flag by test whether last frame was CELP

    isIndepFla g : flag, signalling that the current frame is independent from the previous frame



    [0247] Listed in table 11, the following combinations signalled through flags isTCX10, isTCX20 and isCelpToTCX are allowed with IGF:
    Table 11: TCX transitions, transition factor f, window length n
    Bitrate / ModeisTCX10isTCX20isCelpToTCXTransition factor fWindow length n
    9.6 kbps / WB false true false 1.00 320
    false true true 1.25 400
    9.6 kbps / SWB false true false 1.00 640
    false true true 1.25 800
    13.2 kbps/SWB false true false 1.00 640
    false true true 1.25 800
    16.4 kbps / SWB false true false 1.00 640
    false true true 1.25 800
    24.4 kbps / SWB false true false 1.00 640
    false true true 1.25 800
    32.0 kbps/SWB false true false 1.00 640
    false true true 1.25 800
    48.0 kbps / SWB false true false 1.00 640
    false true true 1.00 640
    true false false 0.50 320
    16.4 kbps / FB false true false 1.00 960
    false true true 1.25 1200
    24.4 kbps / FB false true false 1.00 960
    false true true 1.25 1200
    32.0 kbps / FB false true false 1.00 960
    false true true 1.25 1200
    48.0 kbps / FB false true false 1.00 960
    false true true 1.00 960
    true false false 0.50 480
    96.0 kbps / FB false true false 1.00 960
    false true true 1.00 960
    true false false 0.50 480
    128.0 kbps / FB false true false 1.00 960
    false true true 1.00 960
    true false false 0.50 480

    5.3.3.2.11.3 IGF functions on transmission (TX) side



    [0248] All function declaration assumes that input elements are provided by a frame by frame basis. The only exceptions are two consecutive TCX 10 frames, where the second frame is encoded dependent on the first frame.

    5.3.3.2.11.4 IGF scale factor calculation



    [0249] This subclause describes how the IGF scale factor vector g(k), k = 0,1,..., nB - 1 is calculated on transmission (TX) side.

    5.3.3.2.11.4.1 Complex valued calculation



    [0250] In case the TCX power spectrum P is available the IGF scale factor values g are calculated using P :

    and let m : N → N[be the mapping function which maps the IGF target range into the IGF source range described in subclause 5.3.3.2.11.1.8, calculate:



    where t(0),t(1),...,t(nB) shall be already mapped with the function tF, see subclause 5.3.3.2.11.1.1, and nB are the number of IGF scale factor bands, see table 8.

    [0251] Calculate g(k) with:

    and limit g(k) to the range [0,91] ⊂ Z with



    [0252] The values g(k), k = 0,1,..., nB-1, will be transmitted to the receiver (RX) side after further lossless compression with an arithmetic coder described in subclause 5.3.3.2.11.8.

    5.3.3.2.11.4.2 Real valued calculation



    [0253] If the TCX power spectrum is not available calculate:

    where t(0),t(1),...,t(nB) shall be already mapped with the function tF, see subclause 5.3.3.2.11.1.1, and nB are the number of bands, see table 8.

    [0254] Calculate g(k) with:

    and limit g(k) to the range [0,91]⊂Z with



    [0255] The values g(k), k = 0,1,...,nB-1, will be transmitted to the receiver (RX) side after further lossless compression with an arithmetic coder described in subclause 5.3.3.2.11.8.

    5.3.3.2.11.5 IGF tonal mask



    [0256] In order to determine which spectral components should be transmitted with the core coder, a tonal mask is calculated. Therefore all significant spectral content is identified whereas content that is well suited for parametric coding through IGF is quantized to zero.

    5.3.3.2.11.5.1 IGF tonal mask calculation



    [0257] In case the TCX power spectrum P is not available, all spectral content above t(0) is deleted:

    where R is the real valued TCX spectrum after applying TNS and n is the current TCX window length.

    [0258] In case the TCX power spectrum P is available, calculate:

    where t(0) is the first spectral line in IGF range.

    [0259] Given EHP, apply the following algorithm:


    5.3.3.2.11.6 IGF spectral flatness calculation



    [0260] 
    Table 12: Number of tiles nT and tile width wT
    BitrateModenTwT
    9.6 kbps WB 2 t(2)-t(0),t(nB)-t(2)
    9.6 kbps SWB 3 t(1)-t(0),t(2)-t(1),t(nB)-t(2)
    13.2 kbps SWB 2 t(4)-t(0),t(nB)-t(4)
    16.4 kbps SWB 3 t(4)-t(0),t(6)-t(4),t(nB)-t(6)
    24.4 kbps SWB 3 t(4)-t(0),t(7)-t(4),t(nB)-t(7)
    32.2 kbps SWB 3 t(4)-t(0),t(7)-t(4),t(nB)-t(7)
    48.0 kbps SWB 1 t(nB)-t(0)
    16.4 kbps FB 3 t(4)-t(0),t(7)-t(4),t(nB)-t(7)
    24.4 kbps FB 4 t(4)-t(0),t(6)-t(4),t(9)-t(6),t(nB)-t(9)
    32.0 kbps FB 4 t(4)-t(0),t(6)-t(4),t(9)-t(6),t(nB)-t(9)
    48.0 kbps FB 1 t(nB)-t(0)
    96.0 kbps FB 1 t(nB)-t(0)
    128.0 kbps FB 1 t(nB)-t(0)


    [0261] For the IGF spectral flatness calculation two static arrays, prevFIR and prevIIR, both of size nT are needed to hold filter-states over frames. Additionally a static flag wasTransient is needed to save the information of the input flag isTransient from the previous frame.

    5.3.3.2.11.6.1 Resetting filter states



    [0262] The vectors prevFIR and prevIIR are both static arrays of size nT in the IGF module and both arrays are initialised with zeroes:



    [0263] This initialisation shall be done
    • with codec start up
    • with any bitrate switch
    • with any codec type switch
    • with a transition from CELP to TCX, e.g. isCelpToTCX = true
    • if the current frame has transient properties, e.g. isTransient = true

    5.3.3.2.11.6.2 Resetting current whitening levels



    [0264] The vector currWLevel shall be initialised with zero for all tiles,

    • with codec start up
    • with any bitrate switch
    • with any codec type switch
    • with a transition from CELP to TCX, e.g. isCelpToTCX = true

    5.3.3.2.11.6.3 Calculation of spectral flatness indices



    [0265] The following steps 1) to 4) shall be executed consecutive:
    1. 1) Update previous level buffers and initialize current levels:

      In case prevIsTransient or isTransient is true, apply

      else, if the power spectrum P is available, calculate

      with

      where SFM is a spectral flatness measurement function, described in subclause 5.3.3.2.11.1.3 and CREST is a crest-factor function described in subclause 5.3.3.2.11.1.4.
      Calculate:

      After calculation of the vector s(k), the filter states are updated with:

    2. 2) A mapping function hT : N×P → N is applied to the calculated values to obtain a whitening level index vector currWLevel The mapping function hT: N×P → N is described in subclause 5.3.3.2.11.1.5.

    3. 3) With selected modes, see table 13, apply the following final mapping:

    Table 13: modes for step 4) mapping
    BitrateModemapping
    9.6 kbps WB apply
    9.6 kbps SWB apply
    13.2 kbps SWB NOP
    16.4 kbps SWB apply
    24.4 kbps SWB apply
    32.2 kbps SWB apply
    48.0 kbps SWB NOP
    16.4 kbps FB apply
    24.4 kbps FB apply
    32.0 kbps FB apply
    48.0 kbps FB NOP
    96.0 kbps FB NOP
    128.0 kbps FB NOP


    [0266] After executing step 4) the whitening level index vector currWLevel is ready for transmission.

    5.3.3.2.11.6.4 Coding of IGF whitening levels



    [0267] IGF whitening levels, defined in the vector currWLevel, are transmitted using 1 or 2 bits per tile. The exact number of total bits required depends on the actual values contained in currWLevel and the value of the isIndep flag. The detailed processing is described in the pseudo code below:



    wherein the vector prevWLevel contains the whitening levels from the previous frame and the function encode_whitening_level takes care of the actual mapping of the whitening level currWLevel(k) to a binary code. The function is implemented according to the pseudo code below:


    5.3.3.2.11.7 IGF temporal flatness indicator



    [0268] The temporal envelope of the reconstructed signal by the IGF is flattened on the receiver (RX) side according to the transmitted information on the temporal envelope flatness, which is an IGF flatness indicator.

    [0269] The temporal flatness is measured as the linear prediction gain in the frequency domain. Firstly, the linear prediction of the real part of the current TCX spectrum is performed and then the prediction gain ηigf is calculated:

    where ki = i-th PARCOR coefficient obtained by the linear prediction.

    [0270] From the prediction gain ηigf and the prediction gain ηtns described in subclause 5.3.3.2.2.3, the IGF temporal flatness indicator flag isIgfTemFlat is defined as


    5.3.3.2.11.8 IGF noiseless coding



    [0271] The IGF scale factor vector g is noiseless encoded with an arithmetic coder in order to write an efficient representation of the vector to the bit stream.

    [0272] The module uses the common raw arithmetic encoder functions from the infrastructure, which are provided by the core encoder. The functions used are ari_encode_14bits_sign(bit), which encodes the value bit, ari_encode_14bits_ext(value,cumulativeFrequencyTable), which encodes value from an alphabet of 27 symbols (SYMBOLS_IN_TABLE) using the cumulative frequency table cumulativeFrequencyTable, ari_start_encoding_14bits(), which initializes the arithmetic encoder, and ari_finish_encoding_146its(), which finalizes the arithmetic encoder.

    5.3.3.2.11.8.1 IGF independency flag



    [0273] The internal state of the arithmetic encoder is reset in case the isIndepFlag flag has the value true. This flag may be set to false only in modes where TCX10 windows (see table 11) are used for the second frame of two consecutive TCX 10 frames.

    5.3.3.2.11.8.2 IGF all-Zero flag



    [0274] The IGF all-Zero flag signals that all of the IGF scale factors are zero:



    [0275] The allZero flag is written to the bit stream first. In case the flag is true, the encoder state is reset and no further data is written to the bit stream, otherwise the arithmetic coded scale factor vector g follows in the bit stream.

    5.3.3.2.11.8.3 IGF arithmetic encoding helper functions


    5.3.3.2.11.8.3.1 The reset function



    [0276] The arithmetic encoder states consist of t ∈ {0,1}, and the prev vector, which represents the value of the vector g preserved from the previous frame. When encoding the vector g , the value 0 for t means that there is no previous frame available, therefore prev is undefined and not used. The value 1 for t means that there is a previous frame available therefore prev has valid data and it is used, this being the case only in modes where TCX10 windows (see table 11) are used for the second frame of two consecutive TCX 10 frames. For resetting the arithmetic encoder state, it is enough to set t = 0 .

    [0277] If a frame has isIndepFlag set, the encoder state is reset before encoding the scale factor vector g . Note that the combination t = 0 and isIndepFlag = false is valid, and may happen for the second frame of two consecutive TCX 10 frames, when the first frame had allZero=1. In this particular case, the frame uses no context information from the previous frame (the prev vector), because t = 0, and it is actually encoded as an independent frame.

    5.3.3.2.11.8.3.2 The arith_encode_bits function



    [0278] The arith_encode_bits function encodes an unsigned integer x, of length nBits bits, by writing one bit at a time. arith_encode_bits (x, nBits) { for (i = nBits - 1; i >= 0; --i) { bit = (x >> i) & 1; ari_encode_14bits_sign(bit); } }

    5.3.3.2.11.8.3.2 The save and restore encoder state functions



    [0279] Saving the encoder state is achieved using the function iisIGFSCFEncoderSaveContextState , which copies t and prev vector into tSave and prevSave vector, respectively. Restoring the encoder state is done using the complementary function iisIGFSCFEncoderRestoreContextState, which copies back tSave and prevSave vector into t and prev vector, respectively.

    5.3.3.2.11.8.4 IGF arithmetic encoding



    [0280] Please note that the arithmetic encoder should be capable of counting bits only, e.g., performing arithmetic encoding without writing bits to the bit stream. If the arithmetic encoder is called with a counting request, by using the parameter doRealEncoding set to false , the internal state of the arithmetic encoder shall be saved before the call to the top level function iisIGFSCFEncoderEncode and restored and after the call, by the caller. In this particular case, the bits internally generated by the arithmetic encoder are not written to the bit stream. The arith_encode_residual function encodes the integer valued prediction residual x, using the cumulative frequency table cumulativeFrequencyTable, and the table offset tableOffset. The table offset tableOffset is used to adjust the value x before encoding, in order to minimize the total probability that a very small or a very large value will be encoded using escape coding, which slightly is less efficient. The values which are between MIN_ENC_SEPARATE= -12 and MAX_ENC_SEPARATE= 12, inclusive, are encoded directly using the cumulative frequency table cumulativeFrequencyTable, and an alphabet size of SYMBOLS_IN_TABLE= 27.

    [0281] For the above alphabet of SYMBOLS_IN_TABLE symbols, the values 0 and SYMBOLS_IN_TABLE-1 are reserved as escape codes to indicate that a value is too small or too large to fit in the default interval. In these cases, the value extra indicates the position of the value in one of the tails of the distribution. The value extra is encoded using 4 bits if it is in the range {0,...,14}, or using 4 bits with value 15 followed by extra 6 bits if it is in the range {15 ,...,15+62}, or using 4 bits with value 15 followed by extra 6 bits with value 63 followed by extra 7 bits if it is larger or equal than 15 + 63 . The last of the three cases is mainly useful to avoid the rare situation where a purposely constructed artificial signal may produce an unexpectedly large residual value condition in the encoder. arith_encode_residual (x, cumulativeFrequencyTable, tableOffset) { x += tableOffset; if ((x >= MIN_ENC_SEPARATE) && (x <= MAX_ENC_SEPARATE)) { ari_encode_14bits_ext ((x - MIN_ENC_SEPARATE) + 1, cumulativeFrequencyTable); return; } else if (x < MIN_ENC_SEPARATE) { extra = (MIN_ENC_SEPARATE - 1) - x; ari_encode_14bits_ext (0, cumulativeFrequencyTable); } else { /* x > MAX_ENC_SEPARATE */ extra = x - (MAX_ENC_SEPARATE + 1); ari_encode_14bits_ext(SYMBOLS_IN_TABLE - 1, cumulativeFrequencyTable); } if (extra < 15) { arith_encode_bits(extra, 4); ) else { /* extra >= 15 */ arith_encode_bits(15, 4); extra -= 15; if (extra < 63) { arith encode bits(extra, 6); ) else { /* extra >= 63 */ arith_encode_bits(63, 6); extra = 63; arith_encode_bits(extra, 7); } } }

    [0282] The function encode_sfe_vector encodes the scale factor vector g, which consists of nB integer values. The value t and the prev vector, which constitute the encoder state, are used as additional parameters for the function. Note that the top level function iisIGFSCFEncoderEncode must call the common arithmetic encoder initialization function ari_start_encoding_14bits before calling the function encode_sfe_vector, and also call the arithmetic encoder finalization function ari_done_encoding_14bits afterwards.

    [0283] The function quant_ctx is used to quantize a context value ctx, by limiting it to {-3,...,3}, and it is defined as: quant_ctx(ctx) { if (abs(ctx) <= 3) { return ctx; } else if (ctx > 3) { return 3; ) else { /* ctx < -3 */ return -3; } }

    [0284] The definitions of the symbolic names indicated in the comments from the pseudo code, used for computing the context values, are listed in the following table 14:
    Table 14: Definition of symbolic names
    the previous frame (when available)the current frame
    a = prev[f] x = g[f] (the value to be coded)
    c = prev[f - 1] b = g[f - 1] (when available)
      e = g[f - 2] (when available)
    encode_sfe_vector(t, prev, g, nB) for (f = 0; f < nB; f++) { if (t == 0) { if (f == 0) { ari_encode_14bits_ext(g[f] >> 2, cf_se00); arith_encode_bits (g[f] & 3, 2); /* LSBs as 2 bit raw */ } else if (f == 1) { pred = g[f - 1]; /* pred = b */ arith_encode_residual (g[f] - pred, cf_se01, cf_off_se01); } else { /* f >= 2 */ pred = g[f - 1]; /* pred = b */ ctx = quant_ctx(g[f - 1] - g[f - 2]); /* Q(b - e) */ arith_encode_residual (g[f] - pred, cf_se02[CTX_OFFSET + ctx)], cf_off_se02[IGF_CTX_OFFSET + ctx]); } else { /* t == 1 */ if (f == 0) { pred = prev[f]; /* pred = a */ arith_encode_residual (x[f] - pred, cf_se10, cf_off_se10); } else { /* (t == 1) && (f >= 1) */ pred = prev[f] + g[f - 1] - prev[f - 1]; /* pred = a + b - c */ ctx_f = quant_ctx(prev[f] - prev[f - 1]); /* Q(a - c) */ ctx_t = quant_ctx(g[f - 1] - prev[f - 1]); /* Q(b - c) */ arith_encode_residual (g[f] - pred, cf_se11[CTX_OFFSET + ctx_t] [CTX_OFFSET + ctx_f)], cf_off_se11 [CTX_OFFSET + ctx_t] [CTX_OFFSET + ctx_f]); } } ] }

    [0285] There are five cases in the above function, depending on the value of t and also on the position f of a value in the vector g :
    • when t = 0 and f = 0 , the first scalefactor of an independent frame is coded, by splitting it into the most significant bits which are coded using the cumulative frequency table cf_se00, and the least two significant bits coded directly.
    • when t = 0 and f = 1, the second scale factor of an independent frame is coded (as a prediction residual) using the cumulative frequency table cf_se01.
    • when t = 0 and f ≥ 2 , the third and following scale factors of an independent frame are coded (as prediction residuals) using the cumulative frequency table cf_se02[CTX_OFFSET + ctx], determined by the quantized context value ctx.
    • when t = 1 and f = 0 , the first scalefactor of a dependent frame is coded (as a prediction residual) using the cumulative frequency table cf _se10.
    • when t = 1and f ≥ 1, the second and following scale factors of a dependent frame are coded (as prediction residuals) using the cumulative frequency table cf _se11[CTX_OFFSET + ctx_t][CTX_OFFSET + ctx_f], determined by the quantized context values ctx_t and ctx _f.


    [0286] Please note that the predefined cumulative frequency tables cf _se01, cf _se02, and the table offsets cf _off _se01, cf _off _se02 depend on the current operating point and implicitly on the bitrate, and are selected from the set of available options during initialization of the encoder for each given operating point. The cumulative frequency table cf _se00 is common for all operating points, and cumulative frequency tables cf _se10 and cf _se11, and the corresponding table offsets cf _off _se10 and cf _off _se11 are also common, but they are used only for operating points corresponding to bitrates larger or equal than 48 kbps, in case of dependent TCX 10 frames (when t = 1).

    5.3.3.2.11.9 IGF bit stream writer



    [0287] The arithmetic coded IGF scale factors, the IGF whitening levels and the IGF temporal flatness indicator are consecutively transmitted to the decoder side via bit stream. The coding of the IGF scale factors is described in subclause 5.3.3.2.11.8.4. The IGF whitening levels are encoded as presented in subclause 5.3.3.2.11.6.4. Finally the IGF temporal flatness indicator flag, represented as one bit, is written to the bit stream.

    [0288] In case of a TCX20 frame, i.e. (isTCX 20 = true), and no counting request is signalled to the bit stream writer, the output of the bit stream writer is fed directly to the bit stream. In case of a TCX10 frame ( isTCX10 = true ), where two sub-frames are coded dependently within one 20ms frame, the output of the bit stream writer for each sub-frame is written to a temporary buffer, resulting in a bit stream containing the output of the bit stream writer for the individual sub-frames. The content of this temporary buffer is finally written to the bit stream.


    Claims

    1. Audio encoder for encoding an audio signal having a lower frequency band and an upper frequency band, comprising:

    a detector (802) for detecting a peak spectral region in the upper frequency band of the audio signal;

    a shaper (804) for shaping the lower frequency band using shaping information for the lower band and for shaping the upper frequency band using at least a portion of the shaping information for the lower frequency band, wherein the shaper (804) is configured to additionally attenuate spectral values in the detected peak spectral region in the upper frequency band; and

    a quantizer and coder stage (806) for quantizing a shaped lower frequency band and a shaped upper frequency band and for entropy coding quantized spectral values from the shaped lower frequency band and the shaped upper frequency band.


     
    2. Audio encoder of claim 1, further comprising:

    a linear prediction analyzer (808) for deriving linear prediction coefficients for a time frame of the audio signal by analyzing a block of audio samples in the time frame, the audio samples being band-limited to the lower frequency band,

    wherein the shaper (804) is configured to shape the lower frequency band using the linear prediction coefficients as the shaping information, and

    wherein the shaper (804) is configured to use at least the portion of the linear prediction coefficients derived from the block of audio samples band-limited to the lower frequency band for shaping the upper frequency band in the time frame of the audio signal.


     
    3. Audio encoder of claim 1 or 2, wherein the shaper (804) is configured to calculate a plurality of shaping factors for a plurality of subbands of the lower frequency band using linear prediction coefficients derived from the lower frequency band of the audio signal,
    wherein the shaper (804) is configured to weight, in the lower frequency band, spectral coefficients in a subband of the lower frequency band using a shaping factor calculated for the corresponding subband, and
    to weight spectral coefficients in the upper frequency band using a shaping factor calculated for one of the subbands of the lower frequency band.
     
    4. Audio encoder of claim 3, wherein the shaper (804) is configured to weight the spectral coefficients of the upper frequency band using a shaping factor calculated for a highest subband of the lower frequency band, the highest subband having a highest center frequency among all center frequencies of subbands of the lower frequency band.
     
    5. Audio encoder of one of the preceding claims,
    wherein the detector (802) is configured to determine a peak spectral region in the upper frequency band, when at least one of a group of conditions is true, the group of conditions comprising at least the following:
    a low frequency band amplitude condition (1102), a peak distance condition (1104), and a peak amplitude condition (1106).
     
    6. Audio encoder of claim 5, wherein the detector (802) is configured to determine, for the low-frequency band amplitude condition,
    a maximum spectral amplitude in the lower frequency band (1202);
    a maximum spectral amplitude in the upper frequency band (1204),
    wherein the low frequency band amplitude condition (1102) is true, when the maximum spectral amplitude in the lower frequency band weighted by a predetermined number greater than zero is greater than the maximum spectral amplitude in the upper frequency band (1204).
     
    7. Audio encoder of claim 6,
    wherein the detector (802) is configured to detect the maximum spectral amplitude in the lower frequency band or the maximum spectral amplitude in the upper frequency band before a shaping operation applied by the shaper (804) is applied, or wherein the predetermined number is between 4 and 30.
     
    8. Audio encoder of one of claims 5 to 7,
    wherein the detector (802) is configured to determine, for the peak distance condition,
    a first maximum spectral amplitude in the lower frequency band (1206);
    a first spectral distance of the first maximum spectral amplitude from a border frequency between a center frequency of the lower frequency band (1302) and a center frequency of the upper frequency band (1304);
    a second maximum spectral amplitude in the upper frequency band (1306);
    a second spectral distance of the second maximum spectral amplitude from the border frequency to the second maximum spectral amplitude (1308),
    wherein the peak distance condition (1104) is true, when the first maximum spectral amplitude weighted by the first spectral distance and weighted by a predetermined number being greater than 1 is greater than the second maximum spectral amplitude weighted by the second spectral distance (1310).
     
    9. Audio encoder of claim 8,
    wherein the detector (802) is configured to determine the first maximum spectral amplitude or the second maximum spectral amplitude subsequent to a shaping operation by the shaper (804) without the additional attenuation, or
    wherein the border frequency is the highest frequency in the lower frequency band or the lowest frequency in the upper frequency band, or
    wherein the predetermined number is between 1.5 and 8.
     
    10. Audio encoder of one of claims 5 to 9,
    wherein the detector (802) is configured to determine a first maximum spectral amplitude in a portion of the lower frequency band (1402), the portion extending from a predetermined start frequency of the lower frequency band until a maximum frequency of the lower frequency band, the predetermined start frequency being greater than a minimum frequency of the lower frequency band,
    to determine a second maximum spectral amplitude in the upper frequency band (1404),
    wherein the peak amplitude condition (1106) is true, when the second maximum spectral amplitude is greater than the first maximum spectral amplitude weighted by a predetermined number being greater than or equal to 1 (1406).
     
    11. Audio encoder of claim 10,
    wherein the detector (802) is configured to determine the first maximum spectral amplitude or the second maximum spectral amplitude after a shaping operation applied by the shaper (804) without the additional attenuation, or wherein the predetermined start frequency is at least 10% of the lower frequency band above the minimum frequency of the lower frequency band or wherein the predetermined start frequency is at a frequency being equal to half a maximum frequency of the lower frequency band within a tolerance of plus/minus 10 percent of the half the maximum frequency, or
    wherein the predetermined number depends on a bitrate to be provided by the quantizer/coder stage, so that the predetermined number is higher for a higher bitrate, or
    wherein the predetermined number is between 1.0 and 5.0.
     
    12. Audio encoder of one of claims 6 to 11,
    wherein the detector (802) is configured to determine the peak spectral region only when at least two conditions out of the three conditions or the three conditions are true.
     
    13. Audio encoder of one of claims 6 to 12,
    wherein the detector (802) is configured to determine, as the spectral amplitude, an absolute value of spectral value of the real spectrum, a magnitude of a complex spectrum, any power of the spectral value of the real spectrum or any power of a magnitude of the complex spectrum, the power being greater than 1.
     
    14. Audio encoder of one of the preceding claims,
    wherein the shaper (804) is configured to attenuate at least one spectral value in the detected peak spectral region based on a maximum spectral amplitude in the upper frequency band or based on a maximum spectral amplitude in the lower frequency band.
     
    15. Audio encoder of claim 14,
    wherein the shaper (804) is configured to determine the maximum spectral amplitude in a portion of the lower frequency band, the portion extending from a predetermined start frequency of the lower frequency band until a maximum frequency of the lower frequency band, the predetermined start frequency being greater than a minimum frequency of the lower frequency band, wherein the predetermined start frequency is preferably at least 10% of the lower frequency band above the minimum frequency of the lower frequency band or wherein the predetermined start frequency is preferably at a frequency being equal to half a maximum frequency of the lower frequency band within a tolerance of plus/minus 10 percent of the half the maximum frequency.
     
    16. Audio encoder of one of claims 14 or 15,
    wherein the shaper (804) is configured to additionally attenuate the spectral values using an attenuation factor, the attenuation factor being derived from the maximum spectral amplitude in the lower frequency band (1602) multiplied (1606) by a predetermined number being greater than or equal to 1 and divided by the maximum spectral amplitude in the upper frequency band (1604).
     
    17. Audio encoder of one of the preceding claims,
    wherein the shaper (804) is configured to shape the spectral values in the detected peak spectral region based on:

    a first weighting operation (1702, 804a) using at least the portion of the shaping information for the lower frequency band and a second subsequent weighting operation (1704, 804b) using an attenuation information; or

    a first weighting operation using the attenuation information and a second subsequent weighting information using at least a portion of the shaping information for the lower frequency band, or

    a single weighting operation using a combined weighting information derived from the attenuation information and at least the portion of the shaping information for the lower frequency band.


     
    18. Audio encoder of claim 17,
    wherein the weighting information for the lower frequency band is a set of shaping factors, each shaping factor being associated with a subband of the lower frequency band,
    wherein the at least the portion of the weighting information for the lower frequency band used in the shaping operation for the higher frequency band is a shaping factor associated with a subband of the lower frequency band having a highest center frequency of all subbands in the lower frequency band, or
    wherein the attenuation information is an attenuation factor applied to the at least one spectral value in the detected spectral region or to all the spectral values in the detected spectral region or to all spectral values in the upper frequency band for which the peak spectral region has been detected by the detector (802) for a time frame of the audio signal, or
    wherein the shaper (804) is configured to perform the shaping of the lower and the upper frequency band without any additional attenuation when the detector (802) has not detected any peak spectral region in the upper frequency band of a time frame of the audio signal.
     
    19. Audio encoder of one of the preceding claims,
    wherein the quantizer and coder stage (806) comprises a rate loop processor for estimating a quantizer characteristic so that a predetermined bitrate of an entropy encoded audio signal is obtained.
     
    20. Audio encoder of claim 19, wherein the quantizer characteristic is a global gain,
    wherein the quantizer and coder stage (806) comprises:

    a weighter (1502) for weighting shaped spectral values in the lower frequency band and shaped spectral values in the upper frequency band by the same global gain,

    a quantizer (1504) for quantizing values weighted by the global gain; and

    an entropy coder (1506) for entropy coding the quantized values, wherein the entropy coder comprises an arithmetic coder or an Huffman coder.


     
    21. Audio encoder of one of the preceding claims, further comprising:
    a tonal mask processor (1012) for determining, in the upper frequency band, a first group of spectral values to be quantized and entropy encoded and a second group of spectral values to be parametrically coded by a gap-filling procedure, wherein the tonal mask processor is configured to set the second group of spectral values to zero values.
     
    22. Audio encoder of one of the preceding claims, further comprising:

    a common processor (1002);

    a frequency domain encoder (1012, 802, 804, 806); and

    a linear prediction encoder (1008),

    wherein the frequency domain encoder comprises the detector (802), the shaper (804) and the quantizer and coder stage (806), and

    wherein the common processor is configured calculate data to be used by the frequency domain encoder and the linear prediction encoder.


     
    23. Audio encoder of claim 22,
    wherein the common processor is configured to resample (1006) the audio signal to obtain a resampled audio signal band limited to the lower frequency band for a time frame of the audio signal, and
    wherein the common processor (1002) comprises a linear prediction analyzer (808) for deriving linear prediction coefficients for the time frame of the audio signal by analyzing a block of audio samples in the time frame, the audio samples being band-limited to the lower frequency band, or
    wherein the common processor (1002) is configured to control that the time frame of the audio signal is to be represented by either an output of the linear prediction encoder or an output of the frequency domain encoder.
     
    24. Audio encoder of one of claims 22 to 23,
    wherein the frequency domain encoder comprises a time-to-frequency converter (1012) for converting a time frame of the audio signal into a frequency representation comprising the lower frequency band and the upper frequency band.
     
    25. Method for encoding an audio signal having a lower frequency band and an upper frequency band, comprising:

    detecting (802) a peak spectral region in the upper frequency band of the audio signal;

    shaping (804) the lower frequency band of the audio signal using shaping information for the lower frequency band and shaping (1702) the upper frequency band of the audio signal using at least a portion of the shaping information for the lower frequency band, wherein the shaping of the upper frequency band comprises an additional attenuation (1704) of a spectral value in the detected peak spectral region in the upper frequency band.


     
    26. Computer program for performing, when running on a computer or processor, the method of claim 25.
     


    Ansprüche

    1. Audiocodierer zum Codieren eines Audiosignals, das ein unteres Frequenzband und ein oberes Frequenzband aufweist, mit folgenden Merkmalen:

    einem Detektor (802) zum Detektieren eines Spitzenspektralbereichs in dem oberen Frequenzband des Audiosignals;

    einem Former (804) zum Formen des unteren Frequenzbandes unter Verwendung von Formungsinformationen für das untere Band und zum Formen des oberen Frequenzbandes unter Verwendung zumindest eines Teils der Formungsinformationen für das untere Frequenzband, wobei der Former (804) dazu konfiguriert ist, Spektralwerte in dem detektierten Spitzenspektralbereich in dem oberen Frequenzband zusätzlich zu dämpfen; und

    einer Quantisierer- und Codiererstufe (806) zum Quantisieren eines geformten unteren Frequenzbandes oder eines geformten oberen Frequenzbandes und zum Entropiecodieren quantisierter Spektralwerte ausgehend von dem geformten unteren Frequenzband und dem geformten oberen Frequenzband.


     
    2. Audiocodierer gemäß Anspruch 1, der ferner folgende Merkmale aufweist:

    einen Lineare-Prädiktion-Analysator (808) zum Ableiten von Lineare-Prädiktion-Koeffizienten für einen Zeitrahmen des Audiosignals durch Analysieren eines Blocks von Audioabtastwerten in dem Zeitrahmen, wobei die Audioabtastwerte auf das untere Frequenzband bandbegrenzt sind,

    wobei der Former (804) dazu konfiguriert ist, das untere Frequenzband unter Verwendung der Lineare-Prädiktion-Koeffizienten als Formungsinformationen zu formen, und

    wobei der Former (804) dazu konfiguriert ist, zumindest einen Teil der Lineare-Prädiktion-Koeffizienten, die von dem Block von auf das untere Frequenzband bandbegrenzten Audioabtastwerten abgeleitet sind, zum Formen des oberen Frequenzbandes in dem Zeitrahmen des Audiosignals zu verwenden.


     
    3. Audiocodierer gemäß Anspruch 1 oder 2, bei dem der Former (804) dazu konfiguriert ist, eine Mehrzahl von Formungsfaktoren für eine Mehrzahl von Teilbändern des unteren Frequenzbandes unter Verwendung von Lineare-Prädiktion-Koeffizienten, die von dem unteren Frequenzband des Audiosignals abgeleitet sind, zu berechnen,
    wobei der Formgeber (804) dazu konfiguriert ist, in dem unteren Frequenzband Spektralkoeffizienten in einem Teilband des unteren Frequenzbandes unter Verwendung eines für das entsprechende Teilband berechneten Formungsfaktors zu gewichten, und
    Spektralkoeffizienten in dem oberen Frequenzband unter Verwendung eines für eines der Teilbänder des unteren Frequenzbandes berechneten Formungsfaktors zu gewichten.
     
    4. Audiocodierer gemäß Anspruch 3, bei dem der Former (804) dazu konfiguriert ist, die Spektralkoeffizienten des oberen Frequenzbandes unter Verwendung eines für ein höchstes Teilband des unteren Frequenzbandes berechneten Formungsfaktors zu gewichten, wobei das höchste Teilband eine höchste Mittenfrequenz von allen Mittenfrequenzen von Teilbändern des unteren Frequenzbandes aufweist.
     
    5. Audiocodierer gemäß einem der vorhergehenden Ansprüche,
    bei dem der Detektor (802) dazu konfiguriert ist, einen Spitzenspektralbereich in dem oberen Frequenzband zu ermitteln, wenn zumindest eine einer Gruppe von Bedingungen wahr ist, wobei die Gruppe von Bedingungen zumindest Folgende aufweist:
    eine Niederfrequenzbandamplitudenbedingung (1102), eine Spitzenabstandsbedingung (1104) und eine Spitzenamplitudenbedingung (1106).
     
    6. Audiocodierer gemäß Anspruch 5, bei dem der Detektor (802) dazu konfiguriert ist, für die Niederfrequenzbandamplitudenbedingung Folgendes zu ermitteln:

    eine maximale Spektralamplitude in dem unteren Frequenzband (1202);

    eine maximale Spektralamplitude in dem oberen Frequenzband (1204),

    wobei die Niederfrequenzbandamplitudenbedingung (1102) wahr ist, wenn die maximale Spektralamplitude in dem unteren Frequenzband, die durch eine vorbestimmte Zahl gewichtet wird, die größer ist als null, größer ist als die maximale Spektralamplitude in dem oberen Frequenzband (1204).


     
    7. Audiocodierer gemäß Anspruch 6,
    bei dem der Detektor (802) dazu konfiguriert ist, die maximale Spektralamplitude in dem unteren Frequenzband oder die maximale Spektralamplitude in dem oberen Frequenzband zu detektieren, bevor ein durch den Former (804) angewendeter Formungsvorgang angewendet wird, oder bei dem die vorbestimmte Zahl zwischen 4 und 30 liegt.
     
    8. Audiocodierer gemäß einem der Ansprüche 5 bis 7,
    bei dem der Detektor (802) dazu konfiguriert ist, für die Spitzenabstandsbedingung Folgendes zu ermitteln:

    eine erste maximale Spektralamplitude in dem unteren Frequenzband (1206);

    einen ersten Spektralabstand der ersten maximalen Spektralamplitude von einer Grenzfrequenz zwischen einer Mittenfrequenz des unteren Frequenzbandes (1302) und einer Mittenfrequenz des oberen Frequenzbandes (1304);

    eine zweite maximale Spektralamplitude in dem oberen Frequenzband (1306);

    einen zweiten Spektralabstand der zweiten maximalen Spektralamplitude von der Grenzfrequenz zu der zweiten maximalen Spektralamplitude (1308),

    wobei die Spitzenabstandsbedingung (1104) wahr ist, wenn die erste maximale Spektralamplitude, die durch den ersten Spektralabstand gewichtet wird und durch eine vorbestimmte Zahl gewichtet wird, die größer ist als 1, größer ist als die durch den zweiten Spektralabstand gewichtete zweite maximale Spektralamplitude (1310).


     
    9. Audiocodierer gemäß Anspruch 8,
    bei dem der Detektor (802) dazu konfiguriert ist, die erste maximale Spektralamplitude oder die zweite maximale Spektralamplitude anschließend an einen Formungsvorgang seitens des Formers (804) ohne die zusätzliche Dämpfung zu ermitteln, oder
    bei dem die Grenzfrequenz die höchste Frequenz in dem unteren Frequenzband oder die niedrigste Frequenz in dem oberen Frequenzband ist, oder
    bei dem die vorbestimmte Zahl zwischen 1,5 und 8 liegt.
     
    10. Audiocodierer gemäß einem der Ansprüche 5 bis 9,
    bei dem der Detektor (802) dazu konfiguriert ist, eine erste maximale Spektralamplitude in einem Abschnitt des unteren Frequenzbandes zu ermitteln (1402), wobei sich der Abschnitt von einer vorbestimmten Startfrequenz des unteren Frequenzbandes bis zu einer maximalen Frequenz des unteren Frequenzbandes erstreckt, wobei die vorbestimmte Startfrequenz größer ist als eine minimale Frequenz des unteren Frequenzbandes,
    um eine zweite maximale Spektralamplitude in dem oberen Frequenzband zu ermitteln (1404),
    bei dem die Spitzenamplitudenbedingung (1106) wahr ist, wenn die zweite maximale Spektralamplitude größer ist als die erste maximale Spektralamplitude, die durch eine vorbestimmte Zahl gewichtet wird, die größer als oder gleich 1 ist (1406).
     
    11. Audiocodierer gemäß Anspruch 10,
    bei dem der Detektor (802) dazu konfiguriert ist, die erste maximale Spektralamplitude oder die zweite maximale Spektralamplitude nach einem Formungsvorgang, der durch den Former (804) ohne die zusätzliche Dämpfung angewendet wurde, zu ermitteln, oder bei dem die vorbestimmte Startfrequenz zumindest 10% des unteren Frequenzbandes über der minimalen Frequenz des unteren Frequenzbandes liegt oder bei dem die vorbestimmte Startfrequenz bei einer Frequenz liegt, die der Hälfte einer maximalen Frequenz des unteren Frequenzbandes innerhalb einer Toleranz von plus/minus 10% der Hälfte der maximalen Frequenz entspricht, oder
    bei dem die vorbestimmte Zahl von einer Bitrate abhängt, die durch die Quantisierer/Codierer-Stufe bereitgestellt werden soll, so dass die vorbestimmte Zahl für eine höhere Bitrate höher ist, oder
    bei dem die vorbestimmte Zahl zwischen 1,0 und 5,0 liegt.
     
    12. Audiocodierer gemäß einem der Ansprüche 6 bis 11,
    bei dem der Detektor (802) dazu konfiguriert ist, den Spitzenspektralbereich nur dann zu ermitteln, wenn zumindest zwei Bedingungen der drei Bedingungen oder die drei Bedingungen wahr sind.
     
    13. Audiocodierer gemäß einem der Ansprüche 6 bis 12,
    bei dem der Detektor (802) dazu konfiguriert ist, als Spektralamplitude einen Absolutwert eines Spektralwerts des realen Spektrums, eine Größe eines komplexen Spektrums, eine beliebige Potenz des Spektralwerts des realen Spektrums oder eine beliebige Potenz einer Größe des komplexen Spektrums, wobei die Potenz größer als 1 ist, zu bestimmen.
     
    14. Audiocodierer gemäß einem der vorhergehenden Ansprüche,
    bei dem der Former (804) dazu konfiguriert ist, zumindest einen Spektralwert in dem detektierten Spitzenspektralbereich auf der Basis einer maximalen Spektralamplitude in dem oberen Frequenzband oder auf der Basis einer maximalen Spektralamplitude in dem unteren Frequenzband zu dämpfen.
     
    15. Audiocodierer gemäß Anspruch 14,
    bei dem der Former (804) dazu konfiguriert ist, die maximale Spektralamplitude in einem Abschnitt des unteren Frequenzbandes zu bestimmen, wobei sich der Abschnitt von einer vorbestimmten Startfrequenz des unteren Frequenzbandes bis zu einer maximalen Frequenz des unteren Frequenzbandes erstreckt, wobei die vorbestimmte Startfrequenz größer ist als eine minimale Frequenz des unteren Frequenzbandes, wobei die vorbestimmte Startfrequenz vorzugsweise zumindest 10% des unteren Frequenzbandes über der minimalen Frequenz des unteren Frequenzbandes liegt oder wobei die vorbestimmte Startfrequenz vorzugsweise bei einer Frequenz liegt, die der Hälfte einer maximalen Frequenz des unteren Frequenzbandes innerhalb einer Toleranz von plus/minus 10% der Hälfte der maximalen Frequenz entspricht.
     
    16. Audiocodierer gemäß Anspruch 14 oder 15,
    bei dem der Former (804) dazu konfiguriert ist, die Spektralwerte unter Verwendung eines Dämpfungsfaktors zusätzlich zu dämpfen, wobei der Dämpfungsfaktor von der maximalen Spektralamplitude in dem unteren Frequenzband (1602), multipliziert (1606) mit einer vorbestimmten Zahl, die größer als oder gleich 1 ist, und dividiert durch die maximale Spektralamplitude in dem oberen Frequenzband (1604), abgeleitet ist.
     
    17. Audiocodierer gemäß einem der vorhergehenden Ansprüche,
    bei dem der Former (804) dazu konfiguriert ist, die Spektralwerte in dem detektierten Spitzenspektralbereich auf der Basis der Folgenden zu formen:

    eines ersten Gewichtungsvorgangs (1702, 804a) unter Verwendung zumindest des Teils der Formungsinformationen für das untere Frequenzband, und eines zweiten anschließenden Gewichtungsvorgangs (1704, 804b) unter Verwendung von Dämpfungsinformationen; oder

    eines ersten Gewichtungsvorgangs unter Verwendung der Dämpfungsinformationen und zweiter anschließender Gewichtungsinformationen unter Verwendung zumindest eines Teils der Formungsinformationen für das untere Frequenzband, oder

    eines einzigen Gewichtungsvorgangs unter Verwendung kombinierter Gewichtungsinformationen, die von den Dämpfungsinformationen und zumindest dem Teil der Formungsinformationen für das untere Frequenzband abgeleitet sind.


     
    18. Audiocodierer gemäß Anspruch 17,
    bei dem die Gewichtungsinformationen für das untere Frequenzband ein Satz von Formungsfaktoren sind, wobei jeder Formungsfaktor einem Teilband des unteren Frequenzbandes zugeordnet ist,
    bei dem der zumindest eine Teil der Gewichtungsinformationen für das untere Frequenzband, der bei dem Formungsvorgang für das höhere Frequenzband verwendet wird, ein Formungsfaktor ist, der einem Teilband des unteren Frequenzbandes zugeordnet ist, das eine höchste Mittenfrequenz aller Teilbänder in dem unteren Frequenzband aufweist, oder
    bei dem die Dämpfungsinformationen ein Dämpfungsfaktor sind, der auf den zumindest einen Spektralwert in dem detektierten Spektralbereich oder auf alle Spektralwerte in dem detektierten Spektralbereich oder auf alle Spektralwerte in dem oberen Frequenzband, für das der Spitzenspektralbereich durch den Detektor (802) für einen Zeitrahmen des Audiosignals detektiert wurde, angewendet wird, oder
    bei dem der Former (804) dazu konfiguriert ist, das Formen des unteren und des oberen Frequenzbandes ohne jegliche zusätzliche Dämpfung durchzuführen, wenn der Detektor (802) keinen Spitzenspektralbereich in dem oberen Frequenzband eines Zeitrahmens des Audiosignals detektiert hat.
     
    19. Audiocodierer gemäß einem der vorhergehenden Ansprüche,
    bei dem die Quantisierer- und Codiererstufe (806) einen Ratenschleifenprozessor zum Schätzen einer Quantisierercharakteristik aufweist, so dass eine vorbestimmte Bitrate eines entropiecodierten Audiosignals erhalten wird.
     
    20. Audiocodierer gemäß Anspruch 19, bei dem die Quantisierercharakteristik ein globaler Gewinn ist,
    bei dem die Quantisierer- und Codiererstufe (806) folgende Merkmale aufweist:

    einen Gewichter (1502) zum Gewichten geformter Spektralwerte in dem unteren Frequenzband und geformter Spektralwerte in dem oberen Frequenzband durch denselben globalen Gewinn,

    einen Quantisierer (1504) zum Quantisieren von Werten, die durch den globalen Gewinn gewichtet sind; und

    einen Entropiecodierer (1506) zum Entropiecodieren der quantisierten Werte, wobei der Entropiecodierer einen arithmetischen Codierer oder einen Huffman-Codierer aufweist.


     
    21. Audiocodierer gemäß einem der vorhergehenden Ansprüche, der ferner folgende Merkmale aufweist:
    einen Tonalmaskenprozessor (1012) zum Bestimmen, in dem oberen Frequenzband, einer ersten Gruppe von Spektralwerten, die quantisiert und entropiecodiert werden sollen, und einer zweiten Gruppe von Spektralwerten, die mittels einer Lückenfüllprozedur parametrisch codiert werden sollen, wobei der Tonalmaskenprozessor dazu konfiguriert ist, die zweite Gruppe von Spektralwerten auf Nullwerte einzustellen.
     
    22. Audiocodierer gemäß einem der vorhergehenden Ansprüche, der ferner folgende Merkmale aufweist:

    einen gemeinsamen Prozessor (1002);

    einen Frequenzdomänencodierer (1012, 802, 804, 806); und

    einen Lineare-Prädiktion-Codierer (1008),

    wobei der Frequenzdomänencodierer den Detektor (802), den Former (804) und die Quantisierer- und Codiererstufe (806) aufweist, und

    wobei der gemeinsame Prozessor dazu konfiguriert ist, Daten zu berechnen, die durch den Frequenzdomänencodierer und den Lineare-Prädiktion-Codierer verwendet werden sollen.


     
    23. Audiocodierer gemäß Anspruch 22,
    bei dem der gemeinsame Prozessor dazu konfiguriert ist, das Audiosignal erneut abzutasten (1006), um ein erneut abgetastetes Audiosignal, das auf das untere Frequenzband bandbegrenzt ist, für einen Zeitrahmen des Audiosignals zu erhalten, und
    wobei der gemeinsame Prozessor (1002) einen Lineare-Prädiktion-Analysator (808) zum Ableiten von Lineare-Prädiktion-Koeffizienten für den Zeitrahmen des Audiosignals durch Analysieren eines Blocks von Audioabtastwerten in dem Zeitrahmen aufweist, wobei die Audioabtastwerte auf das untere Frequenzband bandbegrenzt sind, oder
    wobei der gemeinsame Prozessor (1002) dazu konfiguriert ist, zu steuern, dass der Zeitrahmen des Audiosignals entweder durch eine Ausgabe des Lineare-Prädiktion-Codierers oder eine Ausgabe des Frequenzdomänencodierers dargestellt werden soll.
     
    24. Audiocodierer gemäß einem der Ansprüche 22 bis 23,
    bei dem der Frequenzdomänencodierer einen Zeit/Frequenz-Wandler (1012) zum Umwandeln eines Zeitrahmens des Audiosignals in eine Frequenzdarstellung umfasst, die das untere Frequenzband und das obere Frequenzband aufweist.
     
    25. Verfahren zum Codieren eines Audiosignals, das ein unteres Frequenzband und ein oberes Frequenzband aufweist, mit folgenden Schritten:

    Detektieren (802) eines Spitzenspektralbereichs in dem oberen Frequenzband des Audiosignals;

    Formen (804) des unteren Frequenzbandes des Audiosignals unter Verwendung von Formungsinformationen für das untere Frequenzband, und Formen (1702) des oberen Frequenzbandes des Audiosignals unter Verwendung zumindest eines Teils der Formungsinformationen für das untere Frequenzband, wobei das Formen des oberen Frequenzbandes eine zusätzliche Dämpfung (1704) eines Spektralwerts in dem detektierten Spitzenspektralbereich in dem oberen Frequenzband aufweist.


     
    26. Computerprogramm zum Durchführen, wenn es auf einem Computer oder Prozessor abläuft, des Verfahrens gemäß Anspruch 25.
     


    Revendications

    1. Codeur audio pour coder un signal audio présentant une bande de fréquences inférieures et une bande de fréquences supérieures, comprenant:

    un détecteur (802) destiné à détecter une région spectrale de crête dans la bande de fréquences supérieures du signal audio;

    un moyen de mise en forme (804) destiné à mettre en forme la bande de fréquences inférieures à l'aide des informations de mise en forme pour la bande de fréquences inférieures et à mettre en forme la bande de fréquences supérieures à l'aide d'au moins une partie des informations de mise en forme pour la bande de fréquences inférieures, où le moyen de mise en forme (804) est configuré pour atténuer de manière additionnelle les valeurs spectrales dans la région spectrale de crête détectée dans la bande de fréquences supérieures; et

    un étage de quantificateur et codeur (806) destiné à quantifier une bande de fréquences inférieures mise en forme et une bande de fréquences supérieures mise en forme et à coder de manière entropique les valeurs spectrales quantifiées à partir de la bande de fréquences inférieures mise en forme et de la bande de fréquences supérieures mise en forme.


     
    2. Codeur audio selon la revendication 1, comprenant par ailleurs:

    un analyseur de prédiction linéaire (808) destiné à dériver les coefficients de prédiction linéaire pour une trame temporelle du signal audio en analysant un bloc d'échantillons audio dans la trame temporelle, les échantillons audio étant limités en bande à la bande de fréquences inférieures,

    dans lequel le moyen de mise en forme (804) est configuré pour mettre en forme la bande de fréquences inférieures à l'aide des coefficients de prédiction linéaire comme informations de mise en forme, et

    dans lequel le moyen de mise en forme (804) est configuré pour utiliser au moins la partie des coefficients de prédiction linéaire dérivés du bloc d'échantillons audio limité en bande à la bande de fréquences inférieures pour mettre en forme la bande de fréquences supérieures dans la trame temporelle du signal audio.


     
    3. Codeur audio selon la revendication 1 ou 2, dans lequel le moyen de mise en forme (804) est configuré pour calculer une pluralité de facteurs de mise en forme pour une pluralité de sous-bandes de la bande de fréquences inférieures à l'aide des coefficients de prédiction linéaire dérivés de la bande de fréquences inférieures du signal audio,
    dans lequel le moyen de mise en forme (804) est configuré pour pondérer, dans la bande de fréquences inférieures, les coefficients spectraux dans une sous-bande de la bande de fréquences inférieures à l'aide d'un facteur de mise en forme calculé pour la sous-bande correspondante, et
    pour pondérer les coefficients spectraux dans la bande de fréquences supérieures à l'aide d'un facteur de mise en forme calculé pour l'une des sous-bandes de la bande de fréquences inférieures.
     
    4. Codeur audio selon la revendication 3, dans lequel le moyen de mise en forme (804) est configuré pour pondérer les coefficients spectraux de la bande de fréquences supérieures à l'aide d'un facteur de mise en forme calculé pour une sous-bande la plus haute de la bande de fréquences inférieures, la sous-bande la plus haute présentant une fréquence centrale la plus haute parmi toutes les fréquences centrales des sous-bandes de la bande de fréquences inférieures.
     
    5. Codeur audio selon l'une des revendications précédentes,
    dans lequel le détecteur (802) est configuré pour déterminer une région spectrale de crête dans la bande de fréquences supérieures lorsqu'au moins l'une d'un groupe de conditions est vraie, le groupe de conditions comprenant au moins ce qui suit:
    une condition d'amplitude de bande de basses fréquences (1102), une condition de distance de crête (1104) et une condition d'amplitude de crête (1106).
     
    6. Codeur audio selon la revendication 5, dans lequel le détecteur (802) est configuré pour déterminer, pour la condition d'amplitude de bande de basses fréquences,
    une amplitude spectrale maximale dans la bande de fréquences inférieures (1202);
    une amplitude spectrale maximale dans la bande de fréquences supérieures (1204),
    dans lequel la condition d'amplitude de bande de basses fréquences (1102) est vraie lorsque l'amplitude spectrale maximale dans la bande de fréquences inférieures pondérée par un nombre prédéterminé supérieur à zéro est supérieure à l'amplitude spectrale maximale dans la bande de fréquences supérieures (1204).
     
    7. Codeur audio selon la revendication 6,
    dans lequel le détecteur (802) est configuré pour détecter l'amplitude spectrale maximale dans la bande de fréquences inférieures ou l'amplitude spectrale maximale dans la bande de fréquences supérieures avant que ne soit appliquée une opération de mise en forme appliquée par le moyen de mise en forme (804), ou dans lequel le nombre prédéterminé est compris entre 4 et 30.
     
    8. Codeur audio selon l'une des revendications 5 à 7,
    dans lequel le détecteur (802) est configuré pour déterminer, pour la condition de distance de crête,
    une première amplitude spectrale maximale dans la bande de fréquences inférieures (1206);
    une première distance spectrale de la première amplitude spectrale maximale par rapport à une fréquence limite entre une fréquence centrale de la bande de fréquences inférieures (1302) et une fréquence centrale de la bande de fréquences supérieures (1304);
    une deuxième amplitude spectrale maximale dans la bande de fréquences supérieures (1306);
    une deuxième distance spectrale de la deuxième amplitude spectrale maximale de la fréquence limite à la deuxième amplitude spectrale maximale (1308),
    dans lequel la condition de distance de crête (1104) est vraie, lorsque la première amplitude spectrale maximale pondérée par la première distance spectrale et pondérée par un nombre prédéterminé supérieur à 1 est supérieure à la deuxième amplitude spectrale maximale pondérée par la deuxième distance spectrale (1310).
     
    9. Codeur audio selon la revendication 8,
    dans lequel le détecteur (802) est configuré pour déterminer la première amplitude spectrale maximale ou la deuxième amplitude spectrale maximale à la suite d'une opération de mise en forme par le moyen de mise en forme (804) sans l'atténuation additionnelle, ou
    dans lequel la fréquence limite est la fréquence la plus haute dans la bande de fréquences inférieures ou la fréquence la plus basse dans la bande de fréquences supérieures, ou
    dans lequel le nombre prédéterminé est compris entre 1,5 et 8.
     
    10. Codeur audio selon l'une des revendications 5 à 9,
    dans lequel le détecteur (802) est configuré pour déterminer une première amplitude spectrale maximale dans une partie de la bande de fréquences inférieures (1402), la partie s'étendant d'une fréquence de début prédéterminée de la bande de fréquences inférieures jusqu'à une fréquence maximale de la bande de fréquences inférieures, la fréquence de début prédéterminée étant supérieure à une fréquence minimale de la bande de fréquences inférieures,
    pour déterminer une deuxième amplitude spectrale maximale dans la bande de fréquences supérieures (1404),
    dans lequel la condition d'amplitude de crête (1106) est vraie lorsque la deuxième amplitude spectrale maximale est supérieure à la première amplitude spectrale maximale pondérée par un nombre prédéterminé supérieur ou égal à 1 (1406).
     
    11. Codeur audio selon la revendication 10,
    dans lequel le détecteur (802) est configuré pour déterminer la première amplitude spectrale maximale ou la deuxième amplitude spectrale maximale après une opération de mise en forme appliquée par le moyen de mise en forme (804) sans l'atténuation additionnelle, ou dans lequel la fréquence de début prédéterminée est d'au moins 10% de la bande de fréquences inférieures au-dessus de la fréquence minimale de la bande de fréquences inférieures ou dans laquelle la fréquence de début prédéterminée est à une fréquence qui est égale à la moitié d'une fréquence maximale de la bande de fréquences inférieures dans les limites d'une tolérance de plus/moins 10% de la moitié de la fréquence maximale, ou
    dans lequel le nombre prédéterminé dépend d'un taux de bits à fournir par l'étage de quantificateur/codeur, de sorte que le nombre prédéterminé soit supérieur pour un taux de bits supérieur, ou
    dans lequel le nombre prédéterminé est compris entre 1,0 et 5,0.
     
    12. Codeur audio selon l'une des revendications 6 à 11,
    dans lequel le détecteur (802) est configuré pour déterminer la région spectrale de crête uniquement lorsqu'au moins deux conditions parmi les trois conditions, ou les trois conditions, sont vraies.
     
    13. Codeur audio selon l'une des revendications 6 à 12,
    dans lequel le détecteur (802) est configuré pour déterminer, comme amplitude spectrale, une valeur absolue de la valeur spectrale du spectre réel, une amplitude d'un spectre complexe, toute puissance de la valeur spectrale du spectre réel ou toute puissance d'une amplitude du spectre complexe, la puissance étant supérieure à 1.
     
    14. Codeur audio selon l'une des revendications précédentes,
    dans lequel le moyen de mise en forme (804) est configuré pour atténuer au moins une valeur spectrale dans la région spectrale de crête détectée sur base d'une amplitude spectrale maximale dans la bande de fréquences supérieures ou sur base d'une amplitude spectrale maximale dans la bande de fréquences inférieures.
     
    15. Codeur audio selon la revendication 14,
    dans lequel le moyen de mise en forme (804) est configuré pour déterminer l'amplitude spectrale maximale dans une partie de la bande de fréquences inférieures, la partie s'étendant d'une fréquence de début prédéterminée de la bande de fréquences inférieures à une fréquence maximale de la bande de fréquences inférieures, la fréquence de début prédéterminée étant supérieure à une fréquence minimale de la bande de fréquences inférieures, dans lequel la fréquence de début prédéterminée est de préférence au moins de 10% de la bande de fréquences inférieures au-dessus de la fréquence minimale de la bande de fréquences inférieures ou dans lequel la fréquence de début prédéterminée est de préférence à une fréquence égale à la moitié d'une fréquence maximale de la bande de fréquences inférieures dans les limites d'une tolérance de plus/moins 10% de la moitié de la fréquence maximale.
     
    16. Codeur audio selon l'une des revendications 14 ou 15,
    dans lequel le moyen de mise en forme (804) est configuré pour atténuer de manière additionnelle les valeurs spectrales à l'aide d'un facteur d'atténuation, le facteur d'atténuation étant dérivé de l'amplitude spectrale maximale dans la bande de fréquences inférieures (1602) multipliée (1606) par un nombre prédéterminé qui est supérieur ou égal à 1 et divisée par l'amplitude spectrale maximale dans la bande de fréquences supérieures (1604).
     
    17. Codeur audio selon l'une des revendications précédentes,
    dans lequel le moyen de mise en forme (804) est configuré pour mettre en forme les valeurs spectrales dans la région spectrale de crête détectée sur base de:

    une première opération de pondération (1702, 804a) à l'aide d'au moins la partie des informations de mise en forme pour la bande de fréquences inférieures et une deuxième opération de pondération successive (1704, 804b) à l'aide d'une information d'atténuation; ou

    une première opération de pondération à l'aide des informations d'atténuation et une deuxième information de pondération successive à l'aide d'au moins une partie des informations de mise en forme pour la bande de fréquences inférieures, ou

    une opération de pondération unique à l'aide d'une information de pondération combinée dérivée des informations d'atténuation et d'au moins la partie des informations de mise en forme pour la bande de fréquences inférieures.


     
    18. Codeur audio selon la revendication 17,
    dans lequel les informations de pondération pour la bande de fréquences inférieures sont un ensemble de facteurs de mise en forme, chaque facteur de mise en forme étant associé à une sous-bande de la bande de fréquences inférieures,
    dans lequel l'au moins une partie des informations de pondération pour la bande de fréquences inférieures utilisée dans l'opération de mise en forme pour la bande de fréquences supérieures est un facteur de mise en forme associé à une sous-bande de la bande de fréquences inférieures présentant une fréquence centrale la plus haute de toutes les sous-bandes de la bande de fréquences inférieures, ou
    dans lequel les informations d'atténuation sont un facteur d'atténuation appliqué à l'au moins une valeur spectrale dans la région spectrale détectée ou à toutes les valeurs spectrales dans la région spectrale détectée ou à toutes les valeurs spectrales dans la bande de fréquences supérieures pour laquelle la région spectrale de crête a été détectée par le détecteur (802) pour une trame temporelle du signal audio, ou
    dans lequel le moyen de mise en forme (804) est configuré pour effectuer la mise en forme de la bande de fréquences inférieures et de la bande de fréquences supérieures sans aucune atténuation additionnelle lorsque le détecteur (802) n'a pas détecté de région spectrale de crête dans la bande de fréquences supérieures d'une trame temporelle du signal audio.
     
    19. Codeur audio selon l'une des revendications précédentes,
    dans lequel l'étage de quantificateur et codeur (806) comprend un processeur de boucle de taux destiné à estimer une caractéristique de quantificateur de sorte que soit obtenu un taux de bits prédéterminé d'un signal audio codé de manière entropique.
     
    20. Codeur audio selon la revendication 19, dans lequel la caractéristique de quantificateur est un gain global,
    dans lequel l'étage de quantificateur et codeur (806) comprend:

    un pondérateur (1502) destiné à pondérer les valeurs spectrales mises en forme dans la bande de fréquences inférieures et les valeurs spectrales mises en forme dans la bande de fréquences supérieures par le même gain global,

    un quantificateur (1504) destiné à quantifier les valeurs pondérées par le gain global; et

    un codeur entropique (1506) destiné à coder de manière entropique les valeurs quantifiées, où le codeur entropique comprend un codeur arithmétique ou un codeur de Huffman.


     
    21. Codeur audio selon l'une des revendications précédentes, comprenant par ailleurs:
    un processeur de masque tonal (1012) destiné à déterminer, dans la bande de fréquences supérieures, un premier groupe de valeurs spectrales à quantifier et à coder de manière entropique et un deuxième groupe de valeurs spectrales à coder de manière paramétrique par une procédure de remplissage de trous, dans lequel le processeur de masque tonal est configuré pour régler le deuxième groupe de valeurs spectrales à zéro.
     
    22. Codeur audio selon l'une des revendications précédentes, comprenant par ailleurs:

    un processeur commun (1002);

    un codeur dans le domaine de la fréquence (1012, 802, 804, 806); et

    un codeur de prédiction linéaire (1008),

    dans lequel le codeur dans le domaine de la fréquence comprend le détecteur (802), le moyen de mise en forme (804) et l'étage de quantificateur et codeur (806), et

    dans lequel le processeur commun est configuré pour calculer les données à utiliser par le codeur dans le domaine de la fréquence et le codeur de prédiction linéaire.


     
    23. Codeur audio selon la revendication 22,
    dans lequel le processeur commun est configuré pour ré-échantillonner (1006) le signal audio pour obtenir une bande de signal audio ré-échantillonné limitée à la bande de fréquences inférieures pour une trame temporelle du signal audio, et
    dans lequel le processeur commun (1002) comprend un analyseur de prédiction linéaire (808) destiné à dériver les coefficients de prédiction linéaire pour la trame temporelle du signal audio en analysant un bloc d'échantillons audio dans la trame temporelle, les échantillons audio étant limités en bande à la bande de fréquences inférieures, ou
    dans lequel le processeur commun (1002) est configuré pour commander que la trame temporelle du signal audio doit être représentée soit par une sortie du codeur de prédiction linéaire, soit par une sortie du codeur dans le domaine de la fréquence.
     
    24. Codeur audio selon l'une des revendications 22 à 23,
    dans lequel le codeur dans le domaine de la fréquence comprend un convertisseur temps-fréquence (1012) destiné à convertir une trame temporelle du signal audio en une représentation de fréquence comprenant la bande de fréquences inférieures et la bande de fréquences supérieures.
     
    25. Procédé de codage d'un signal audio présentant une bande de fréquences inférieures et une bande de fréquences supérieures, comprenant le fait de:

    détecter (802) une région spectrale de crête dans la bande de fréquences supérieures du signal audio;

    mettre en forme (804) la bande de fréquences inférieures du signal audio à l'aide des informations de mise en forme pour la bande de fréquences inférieures et mettre en forme (1702) la bande de fréquences supérieures du signal audio à l'aide d'au moins une partie des informations de mise en forme pour la bande de fréquences inférieures, où la mise en forme de la bande de fréquences supérieures comprend une atténuation additionnelle (1704) d'une valeur spectrale dans la région spectrale de crête détectée dans la bande de fréquences supérieures.


     
    26. Programme d'ordinateur pour réaliser, lorsqu'il est exécuté sur un ordinateur ou un processeur, le procédé selon la revendication 25.
     




    Drawing













































    REFERENCES CITED IN THE DESCRIPTION



    This list of references cited by the applicant is for the reader's convenience only. It does not form part of the European patent document. Even though great care has been taken in compiling the references, errors or omissions cannot be excluded and the EPO disclaims all liability in this regard.

    Patent documents cited in the description




    Non-patent literature cited in the description