Classifications

• • G—PHYSICS
  • G10—MUSICAL INSTRUMENTS; ACOUSTICS
  • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
  • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
  • G10L19/02—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using spectral analysis, e.g. transform vocoders or subband vocoders
  • G10L19/022—Blocking, i.e. grouping of samples in time; Choice of analysis windows; Overlap factoring
  • G10L19/025—Detection of transients or attacks for time/frequency resolution switching
• • G—PHYSICS
  • G10—MUSICAL INSTRUMENTS; ACOUSTICS
  • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
  • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
  • G10L19/04—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using predictive techniques
  • G10L19/06—Determination or coding of the spectral characteristics, e.g. of the short-term prediction coefficients
• • G—PHYSICS
  • G10—MUSICAL INSTRUMENTS; ACOUSTICS
  • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
  • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
  • G10L19/02—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using spectral analysis, e.g. transform vocoders or subband vocoders
  • G10L19/0204—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using spectral analysis, e.g. transform vocoders or subband vocoders using subband decomposition
  • G10L19/0208—Subband vocoders
• • G—PHYSICS
  • G10—MUSICAL INSTRUMENTS; ACOUSTICS
  • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
  • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
  • G10L19/02—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using spectral analysis, e.g. transform vocoders or subband vocoders
  • G10L19/022—Blocking, i.e. grouping of samples in time; Choice of analysis windows; Overlap factoring
• • G—PHYSICS
  • G10—MUSICAL INSTRUMENTS; ACOUSTICS
  • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
  • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
  • G10L19/02—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using spectral analysis, e.g. transform vocoders or subband vocoders
  • G10L19/032—Quantisation or dequantisation of spectral components
  • G10L19/035—Scalar quantisation
• • G—PHYSICS
  • G10—MUSICAL INSTRUMENTS; ACOUSTICS
  • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
  • G10L21/00—Speech or voice signal processing techniques to produce another audible or non-audible signal, e.g. visual or tactile, in order to modify its quality or its intelligibility
  • G10L21/02—Speech enhancement, e.g. noise reduction or echo cancellation
  • G10L21/038—Speech enhancement, e.g. noise reduction or echo cancellation using band spreading techniques
• • G—PHYSICS
  • G10—MUSICAL INSTRUMENTS; ACOUSTICS
  • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
  • G10L25/00—Speech or voice analysis techniques not restricted to a single one of groups G10L15/00 - G10L21/00
  • G10L25/03—Speech or voice analysis techniques not restricted to a single one of groups G10L15/00 - G10L21/00 characterised by the type of extracted parameters
  • G10L25/18—Speech or voice analysis techniques not restricted to a single one of groups G10L15/00 - G10L21/00 characterised by the type of extracted parameters the extracted parameters being spectral information of each sub-band

Definitions

• the present invention relates to a new method and apparatus for efficient coding of spectral envelopes m audio coding systems.
• the method may be used both for natural audio coding and speech coding and is especially suited for coders using SBR [WO 98/57436] or other high frequency reconstruction methods.
• Audio source coding techniques can be divided into two classes: natural audio coding and speech coding.
• Natural audio coding is commonly used for music or arbitrary signals at medium bitrates, and generally offers wide audio bandwidth. Speech coders are basically limited to speech reproduction but can on the other hand be used at very low bitrates, albeit with low audio bandwidth.
• the signal is generally separated into two major signal components, the "spectral envelope” and the corresponding "residual” signal.
• the term “spectral envelope” refers to the coarse spectral distribution of the signal m a general sense, e.g.
• filter coefficients m an linear prediction based coder or a set of time-frequency averages of subband samples m a subband coder.
• residual refers to the fine spectral distribution in a general sense, e.g. the LPC error signal or subband samples normalized using the above time-frequency averages.
• envelope data refers to the quantized and coded spectral envelope, and "residual data" to the quantized and coded residual. At medium and high bitrates, the residual data constitutes the main part of the bitstream. At very low bitrates, the envelope data constitutes a larger part of the bitstream Hence, it is indeed important to represent the spectral envelope compactly when using lower bitrates.
• P ⁇ or art audio coders and most speech coders use constant length, relatively short, time segments in the generation of envelope data to achieve good temporal resolution.
• this prevents optimal utilisation of the frequency domain masking known from psycho-acoustics.
• modem audio coders employ adaptive window switching, i.e. they switch time segment lengths depending on the signals statistics.
• Clearly a minimum usage of the short segments is a prerequisite for maximum coding gam.
• long transition windows are needed to alter the segment lengths, limiting the switching flexibility.
• the spectral envelope is a function of two variables: time and frequency.
• the encodmg can be done by exploiting redundancy m either direction of the time/frequency plane.
• codmg of the spectral envelope is performed in the frequency direction, using delta coding (DPCM) or vector quantization (VQ).
• the present invention provides a new method, and an apparatus for spectral envelope coding.
• the codmg scheme is designed to meet the special requirements of systems, where the residual signal withm certain frequency regions is excluded from the transmitted data. Examples are systems employing HFR (High Frequency Reconstruction), in particular SBR (Spectral Band Replication), or paramet ⁇ c coders.
• HFR High Frequency Reconstruction
• SBR Spectral Band Replication
• paramet ⁇ c coders Spectral Band Replication
• non-uniform time and frequency sampling of the spectral envelope is obtained by adaptively grouping subband samples from a fixed size filterbank, into frequency bands and time segments, each of which generates one envelope sample. This allows instantaneous selection of arbitrary time and frequency resolution withm the limits of the filterbank. The system defaults to long time segments and high frequency resolution.
• variable time/frequency resolution method is also applicable on envelope encoding based on prediction. Instead of grouping of subband samples, predictor coefficients are generated for time segments of varying lengths according to the system.
• the invention desc ⁇ bes two schemes for signalling of the time and frequency resolution used.
• the first scheme allows arbitrary selection, by explicit signalling of time segment borders and frequency resolutions. In order to reduce the signalling overhead, four classes of granules are used, offe ⁇ ng different cost/flexibility tradeoffs.
• the second scheme exploits the property of a typical programme mate ⁇ al, that transients are separated at least by a time T nm ⁇ n , in order to reduce the number of control bits further.
• the position withm the interval is encoded and sent to the decoder.
• the encoder and decoder share rules that specify the time/frequency distribution of the spectral envelope samples, given a certain combination of subsequent control signals, ensuring an unambiguous decoding of the envelope data.
• the present invention presents a new and efficient method for scalefactor redundancy coding.
• a dirac pulse in the time domain transforms to a constant in the frequency domain, and a dirac in the frequency domain, i.e. a single sinusoid, corresponds to a signal with constant magnitude m the time domain.
• Figs, la - lb illustrate uniform respective non-uniform sampling in time of the spectral envelope.
• Figs. 2a - 2b define, and illustrate usage of four classes of granules.
• Figs. 3a - 3b are two examples of granules, and the corresponding control signals.
• Figs. 4a - 4c illustrate the position signalling system.
• Fig. 5 illustrates time/frequency switched delta coding.
• Fig. 6 is a block diagram of an encoder using the envelope coding according to the invention.
• Fig. 7 is a block diagram of a decoder using the envelope coding according to the invention.
• Fig. 1 shows the time/frequency representation of a musical signal where sustained chords are combined with sharp transients with mamly high frequency contents.
• the chords In the lowband the chords have high power and the transient power is low, whereas the opposite is true m the highband.
• the envelope data that is generated du ⁇ ng time intervals where transients are present is dominated by the high intermittent transient power.
• the spectral envelope of the transposed signal is estimated using the same instantaneous time- /frequency resolution as used for the analysis of the onginal highband. An equalization of the transposed signal is then performed, based on dissimila ⁇ ties in the spectral envelopes. E.g.
• amplification factors m an envelope adjusting filterbank are calculated as the square root of the quotients between o ⁇ gmal signal and transposed signal average power.
• a problem a ⁇ ses The transposed signal has the same "chord-to-transient" power ratio as the lowband. The gams needed in order to adjust the transposed transients to the correct level thus cause the transposed chords to be amplified relative to the o ⁇ gmal highband level for the full duration of the envelope data containing transient energy. These momenta ⁇ ly too loud chord fragments are perceived as pre- and post echoes to the transient, see Fig. la.
• the solution is to maintain a low update rate du ⁇ ng tonal passages, which make up the major parts of a typical programme mate ⁇ al, and by means of a transient detector localize the transient positions, and update the envelope data close to the leading flanks, see Fig lb.
• the update rate is momenta ⁇ ly increased in a time interval after the transient start. This eliminates gam induced post-echoes.
• the time segmenting du ⁇ ng the decay is not as crucial as finding the start of the transient, as will be explained later.
• a non- uniform sampling m time and frequency as outlined above is applicable both on filterbank- and linear prediction-based envelope coding. Different predictor orders may be used for transient and quasi- stationary (tonal) segments.
• frequency resolution refers to a specific set of frequency bands, LPC coefficients or similar, used in the envelope estimate for a particular time segment.
• high frequency resolution or high time resolution can be obtained instantaneously.
• all practical codec bitstreams comprise data pe ⁇ ods, each of which corresponds to a short time segment of the input signal.
• the time segment associated with such a data pe ⁇ od is hereinafter referred to as a "granule”.
• Typical coders use granules of fixed length. The presence of granule bounda ⁇ es imposes constraints on the design of the time segments used for envelope estimation.
• the algo ⁇ thm that generates these time segments may state that a segment "border" is required at a particular location, and that the subsequent segment should have a certain length. However, if a granule boundary falls withm this interval due to fixed length granules, the segment must be split into two parts. This has two implications: First, the number of segments to encode increases, possibly increasing the amount of data to transmit. Second, forced borders may generate segments that are too short for reliable average power estimates. In order to avoid those shortcomings, the present invention uses va ⁇ able length granules. This requires look-ahead in the encoder, as well as extra buffe ⁇ ng the decoder.
• g ⁇ d denote the time segments and the corresponding frequency resolutions to use for a particular signal
• local g ⁇ d denote the g ⁇ d of one granule.
• the g ⁇ d must be signalled to the decoder for correct decoding of the envelope samples.
• m low bitrate applications the number of bits for this "control signal” must be kept at a minimum.
• Two signalling schemes are proposed in the present invention. P ⁇ or to desc ⁇ bmg them m detail, a “baseline system” and some design c ⁇ te ⁇ a are established.
• the time quantization step for the spectral envelope be T q .
• Those steps may be viewed as "subgranules", which are grouped into the aforementioned time segments.
• a granule comp ⁇ ses of 5 subgranules, where S vanes from granule to granule.
• the number of possible segment combinations withm a granule, ranging from one segment for the entire granule to S segments, is given by
• An arbitrary subdivision of the granule can be signalled by S - 1 bits, representing the consecutive subgranules, stating whether a leading segment border is present at the corresponding subgranule or not. (The first and last granule borders need not be signalled here.) Since S is va ⁇ able it must be signalled, and if this scheme is combined with a fixed length granule lowband codec, the position relative the constant length granules must be signalled as well.
• the segment frequency resolutions can be signalled with dynamically allocated control bits, e.g.
• the minimum time-span between consecutive transients m music programme mate ⁇ al can be estimated in the following way:
• the rhythmic "pulse" is desc ⁇ bed by a time signature expressed as a fraction AIB, where A denotes the number of "beats" per bar and XIB is the type of note corresponding to one beat, for example a 1/4 note, commonly referred to as a quarter note.
• t denote the tempo in Beats Per Mmute (BPM)
• BPM Beats Per Mmute
• T q The necessary time resolution T q must also be established.
• a transient signal has its mam energy in the highband to be reconstructed. This means that the encoded spectral envelope must carry all the "timing" information. The desired timing precision thus determines the resolution needed for encoding of leading flanks.
• T q is much smaller than the minimum note period T nm ⁇ n , since small time deviations withm the pe ⁇ od clearly can be heard.
• the transient has significant energy in the lowband.
• the above desc ⁇ bed gam-induced pre-echoes must fall withm the so called pre- or backward masking time T m of the human auditory system m order to be inaudible.
• T q must satisfy two conditions:
• T m ⁇ T nm ⁇ n (otherwise the notes would be so fast that they could not be resolved) and according to ["Modeling the Additivity of Nonsimultaneous Masking", Hea ⁇ ng Res., vol. 80, pp. 105- 118 (1994)], T m amounts to 10-20 ms. Since T nm ⁇ n is in the 50ms range, a reasonable selection of T q according to Eq 3 results in that the second condition is also met. Of course the precision of the transient detection m the encoder and the time resolution of the analysis/synthesis filterbank must also be considered when selecting T q . Tracking of trailing flanks is less crucial, for several reasons: First, the note-off position has little or no effect on the perceived rhythm. Second, most instruments do not exhibit sharp trailing flanks, but rather a smooth decay curve, i.e. a well defined note-off time does not exist. Third, the post- or forward masking time is substantially longer than the pre-maskmg time.
• both systems according to the present invention employ two time sampling modes; uniform and non-uniform sampling in time.
• the uniform mode is used du ⁇ ng quasi-stationary passages, whereby fixed length segments are used, and little extra signalling is required.
• the system switches to non-uniform operation and granules of va ⁇ able length are used, enabling a good fit to the ideal global g ⁇ d.
• the granules are divided into four classes, and the control signals are tailored towards the specific needs of each class.
• the classes are defined m Fig. 2a.
• Class “FixFix” corresponds to conventional constant length granules
• Class “FixVar” has a movable stop boundary, which allows the granule length to vary.
• Class “VarFix” has a va ⁇ able start boundary, whereas the stop border is fixed.
• the last class. "VarVar” has variable boundaries at both ends. All va ⁇ able boundaries can be offset -a / +b versus the "nominal positions”.
• Fig 2b gives an example of a sequence of granules.
• the system defaults to class FixFix.
• a transient detector (or psycho-acoustical model) operates on a time region ahead of the current granule, as outlined in the figure.
• a class FixVar granule is used - the system switches from uniform to non-uniform operation.
• this granule is followed by a class VarFix granule, since transients most of the time are separated by a number of granules for all practical selections of granule lengths.
• the VarVar class frames may be used.
• Fig 3a is an example of a class FixVar - VarFix pair, and the corresponding control signal.
• One transient is present, and the leading flank (quantized to T q ) is denoted by t.
• the first part of the bitstream is the "class" signal. Since four classes are used, two bits are used for this signal.
• the next signal desc ⁇ bes the location of the va ⁇ able boundary, expressed as the offset from the nominal position. This boundary is referred to as the "absolute border”.
• the segment borders withm the granules are desc ⁇ bed by means of "relative borders": The absolute border is used as a reference, and the other borders are desc ⁇ bed as cumulative distances to the reference.
• the number of relative borders is va ⁇ able, and is signalled to the decoder, after the absolute border.
• a zero number means that the granule comp ⁇ ses one time segment only.
• the segment lengths are signalled in a reversed sequence, moving away from the absolute border at the end of the granule.
• the length of the first segment m a FixVar granule is de ⁇ ved from the relative borders and the total length, and is not signalled.
• Class VarFix relative border signals are inserted into the bitsream m a forward sequence, whereby the last segment length is excluded.
• the bitstream signal order is identical to that of class FixVar, that is: [class, abs. border, number of rel. borders, rel. border 0, rel. border 1 , ... , rel. border N- X]
• the signals are shown in "clear text" instead of the actual binary code words sent m the bitstream.
• Fig 3b shows an alternative coding of the signal.
• the va ⁇ able boundary offers versatility when grouping the segments at a given global g ⁇ d.
• some payload control can be performed at this level, e.g. to equalize the number of bits per granule. This may ease the operation of the lowband encoder.
• Given enough look-ahead, a multipass encoding can be performed, and the optimum combination of local g ⁇ ds be used.
• the absolute border in addition to the above function, serves to align a group of borders around the transient with the precision T q .
• the highest precision is always available for coding of transient leading flanks, and a coarser resolution is used in the tracking of the decay.
• the VarVar class frames use a combination of the FixVar and VarFix signalling, e.g. interleaved: [class, abs. bord. left, d:o ⁇ ght, num. rel. bord left, d:o right, [rel. bord. left 0,..., rel. bord. left N - X , [d:o ⁇ ght]].
• This class offers the greatest flexibility m the local g ⁇ d selection, at the cost of an increased signalling overhead.
• the FixFix class does not require other signals than the class signal per se, m which case for example two (equal length) segments are used. However, it is feasible to add a signal that enables selection withm a set of predefined g ⁇ ds.
• the spectral envelope can be calculated for two segments, and if the two envelopes do not differ more than a certain amount, only one set of envelope data is sent. So far, only the segmenting m time has been desc ⁇ bed. For many reasons, it may be desirable to signal to the decoder which of the borders that corresponds to a transient leading edge. This can be accomplished by sending a "pointer" that points to the relevant border. The reference direction can follow that of the relative borders, and a zero value imply that no transient start is present within the current granule. Furthermore, the frequency resolution (number of power estimates or predictor order) used for the individual segments must also be defined. This can be signalled exphcitely, as m the "baseline system", or implicitely, i.e. the resolution is coupled to the segment lengths, and possibly the pointer position.
• the second system hereinafter referred to as the "position-signalling system" is intended for very low bitrate applications.
• the previously established design rules are used to a greater extent, in order to reduce the number of control signal bits even further.
• a transient detector operating on intervals of length N, located Ni l ahead of the current granule, is employed, Fig. 4b
• a flag associated with this region is set.
• the transient detector has detected a transient in subgranule 2 at time n - X, and a transient m subgranule 3 at time n.
• These positions, pos(n - 1) and pos( ), as well as the corresponding flags, 7 ⁇ g( « - 1) are used as input to the g ⁇ d generation algo ⁇ thm, and the corresponding local g ⁇ d for granule n might be as shown in Fig. 4c.
• subgranule 3 of the granule at time n - 1 is included m the time/frequency g ⁇ d of granule n.
• the only signals fed to the bitstream, are flag(n) [1 bit], and pos(n) [ce ⁇ l( ln_ (N )) bits] .
• the g ⁇ d algo ⁇ thm is also known by the decoder, hence those signals, together with the corresponding signals of the preceding granule n - 1, are sufficient for unambiguous reconstruction of the g ⁇ d used by the encoder.
• the position signal is obsolete, and can be replaced, for example by a 1 bit signal, stating whether one or two segments are used.
• uniform mode operation is identical to that of the class signalling system.
• This system may be viewed as a finite state machine, where the above desc ⁇ bed signals control the transitions from state to state, and the states define the local g ⁇ ds.
• the states can be represented by tables, stored in both the encoder, and the decoder. Since the g ⁇ ds are hard coded, the ability to adaptively alter the payload has been sac ⁇ ficed. A reasonable approach is to keep the time/frequency data mat ⁇ x size (e.g. number of power estimates) approximately constant. Assuming that the number of scalefactors or coefficients m a high resolution segment is two times that of a low resolution segment, one high resolution segment can be traded for two low resolution segments.
• Time/Frequency Switched Scalefactor Encoding Utilizing a time to frequency transform it can be shown that a pulse m the time domain corresponds to a flat spectrum in the frequency domain, and a "pulse" in the frequency domain, i.e. a single sinusoidal, corresponds to a quasi-stationary signal m the time domain. In other words a signal usually shows more transient properties in one domain than the other. In a spectrogram, l e. a time/frequency mat ⁇ x display, this property is evident, and can advantageously be used when coding spectral envelopes.
• a tonal stationary signal can have a very sparse spectrum not suitable for delta codmg in the frequency- direction, but well suited for delta coding m the time -direction, and vice versa. This is displayed in Fig.
• T/F-codmg a time/frequency switching method, hereinafter referred to as T/F-codmg:
• the scalefactors are quantized and coded both in the time- and frequency-direction. For both cases, the required number of bits is calculated for a given coding error, or the error is calculated for a given number of bits. Based upon this, the most beneficial coding direction is selected.
• DPCM and Huffman redundancy coding can be used. Two vectors are calculated, Df and D t :
• Start values are transmitted whenever the spectral envelope is coded in the frequency direction but not when coded in the time direction since they are available at the decoder, through the previous envelope.
• the proposed algo ⁇ thm also require extra information to be transmitted, namely a time/frequency flag indicating in which direction the spectral envelope was coded.
• the T F algo ⁇ thm can advantageously be used with several different coding schemes of the scalefactor-envelope representation apart from DPCM and Huffman, such as ADPCM, LPC and vector quantisation
• the proposed T/F algo ⁇ thm gives significant bitrate-reduction for the spectral-envelope data.
• the analogue input signal is fed to an A D-converter 601, forming a digital signal.
• the digital audio signal is fed to a perceptual audio encoder 602, where source coding is performed.
• the digital signal is fed to a transient detector 603 and to an analysis filterbank 604, which splits the signal into its spectral equivalents (subband signals).
• the transient detector could operate on the subband signals from the analysis bank, but for generality purposes it is here assumed to operate on the digital time domain samples directly.
• the transient detector divides the signal into granules and determines, according to the invention, whether subgranules within the granules is to be flagged as transient.
• This information is sent to the envelope grouping block 605, which specifies the time/frequency grid to be used for the current granule.
• the block combines the uniform sampled subband signals, to form the non-uniform sampled envelope values.
• these values may represent the average power density of the grouped subband samples.
• the envelope values are, together with the grouping information, fed to the envelope encoder block 606.
• This block decides in which direction (time or frequency) to encode the envelope values.
• the resulting signals, the output from the audio encoder, the wideband envelope information, and the control signals are fed to the multiplexer 607, forming a se ⁇ al bitstream that is transmitted or stored.
• the decoder side of the invention is shown in Fig.
• the demultiplexer 701 restores the signals and feeds the approp ⁇ ate part to an audio decoder 702, which produces a low band digital audio signal.
• the envelope information is fed from the demultiplexer to the envelope decoding block 703, which, by use of control data, determines m which direction the current envelope are coded and decodes the data.
• the low band signal from the audio decoder is routed to the transposition module 704, which generates a replicated high band signal from the low band.
• the high band signal is fed to an analysis filterbank 706, which is of the same type as on the encoder side.
• the subband signals are combined in the scalefactor grouping unit 707.
• the same type of combination and time/frequency dist ⁇ bution of the subband samples is adopted as on the encoder side.
• the envelope information from the demultiplexer and the information from the scalefactor grouping unit is processed in the gam control module 708.
• the module computes gam factors to be applied to the subband samples before recombination in the synthesis filterbank block 709.
• the output from the synthesis filterbank is thus an envelope adjusted high band audio signal.
• This signal is added to the output from the delay unit 705, which is fed with the low band audio signal. The delay compensates for the processing time of the high band signal.
• the obtained digital wideband signal is converted to an analogue audio signal in the digital to analogue converter 710.

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