EP3327722B1 - Extension améliorée de bande de fréquence dans un décodeur de signaux audiofréquences - Google Patents

Extension améliorée de bande de fréquence dans un décodeur de signaux audiofréquences Download PDF

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EP3327722B1
EP3327722B1 EP17206569.0A EP17206569A EP3327722B1 EP 3327722 B1 EP3327722 B1 EP 3327722B1 EP 17206569 A EP17206569 A EP 17206569A EP 3327722 B1 EP3327722 B1 EP 3327722B1
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signal
band
frequency
khz
low
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EP3327722A1 (fr
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Magdalena KANIEWSKA
Stéphane RAGOT
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Koninklijke Philips NV
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    • 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
    • B—PERFORMING OPERATIONS; TRANSPORTING
    • B41—PRINTING; LINING MACHINES; TYPEWRITERS; STAMPS
    • B41K—STAMPS; STAMPING OR NUMBERING APPARATUS OR DEVICES
    • B41K3/00—Apparatus for stamping articles having integral means for supporting the articles to be stamped
    • B41K3/54—Inking devices
    • B41K3/56—Inking devices using inking pads
    • B—PERFORMING OPERATIONS; TRANSPORTING
    • B41—PRINTING; LINING MACHINES; TYPEWRITERS; STAMPS
    • B41K—STAMPS; STAMPING OR NUMBERING APPARATUS OR DEVICES
    • B41K1/00—Portable hand-operated devices without means for supporting or locating the articles to be stamped, i.e. hand stamps; Inking devices or other accessories therefor
    • B41K1/02—Portable hand-operated devices without means for supporting or locating the articles to be stamped, i.e. hand stamps; Inking devices or other accessories therefor with one or more flat stamping surfaces having fixed images
    • B41K1/04—Portable hand-operated devices without means for supporting or locating the articles to be stamped, i.e. hand stamps; Inking devices or other accessories therefor with one or more flat stamping surfaces having fixed images with multiple stamping surfaces; with stamping surfaces replaceable as a whole
    • B—PERFORMING OPERATIONS; TRANSPORTING
    • B41—PRINTING; LINING MACHINES; TYPEWRITERS; STAMPS
    • B41K—STAMPS; STAMPING OR NUMBERING APPARATUS OR DEVICES
    • B41K1/00—Portable hand-operated devices without means for supporting or locating the articles to be stamped, i.e. hand stamps; Inking devices or other accessories therefor
    • B41K1/08—Portable hand-operated devices without means for supporting or locating the articles to be stamped, i.e. hand stamps; Inking devices or other accessories therefor with a flat stamping surface and changeable characters
    • B41K1/10—Portable hand-operated devices without means for supporting or locating the articles to be stamped, i.e. hand stamps; Inking devices or other accessories therefor with a flat stamping surface and changeable characters having movable type-carrying bands or chains
    • B—PERFORMING OPERATIONS; TRANSPORTING
    • B41—PRINTING; LINING MACHINES; TYPEWRITERS; STAMPS
    • B41K—STAMPS; STAMPING OR NUMBERING APPARATUS OR DEVICES
    • B41K1/00—Portable hand-operated devices without means for supporting or locating the articles to be stamped, i.e. hand stamps; Inking devices or other accessories therefor
    • B41K1/08—Portable hand-operated devices without means for supporting or locating the articles to be stamped, i.e. hand stamps; Inking devices or other accessories therefor with a flat stamping surface and changeable characters
    • B41K1/12—Portable hand-operated devices without means for supporting or locating the articles to be stamped, i.e. hand stamps; Inking devices or other accessories therefor with a flat stamping surface and changeable characters having adjustable type-carrying wheels
    • B—PERFORMING OPERATIONS; TRANSPORTING
    • B41—PRINTING; LINING MACHINES; TYPEWRITERS; STAMPS
    • B41K—STAMPS; STAMPING OR NUMBERING APPARATUS OR DEVICES
    • B41K1/00—Portable hand-operated devices without means for supporting or locating the articles to be stamped, i.e. hand stamps; Inking devices or other accessories therefor
    • B41K1/36—Details
    • B41K1/38—Inking devices; Stamping surfaces
    • B—PERFORMING OPERATIONS; TRANSPORTING
    • B41—PRINTING; LINING MACHINES; TYPEWRITERS; STAMPS
    • B41K—STAMPS; STAMPING OR NUMBERING APPARATUS OR DEVICES
    • B41K1/00—Portable hand-operated devices without means for supporting or locating the articles to be stamped, i.e. hand stamps; Inking devices or other accessories therefor
    • B41K1/36—Details
    • B41K1/38—Inking devices; Stamping surfaces
    • B41K1/40—Inking devices operated by stamping movement
    • B—PERFORMING OPERATIONS; TRANSPORTING
    • B41—PRINTING; LINING MACHINES; TYPEWRITERS; STAMPS
    • B41K—STAMPS; STAMPING OR NUMBERING APPARATUS OR DEVICES
    • B41K1/00—Portable hand-operated devices without means for supporting or locating the articles to be stamped, i.e. hand stamps; Inking devices or other accessories therefor
    • B41K1/36—Details
    • B41K1/38—Inking devices; Stamping surfaces
    • B41K1/40—Inking devices operated by stamping movement
    • B41K1/42—Inking devices operated by stamping movement with pads or rollers movable for inking
    • 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
    • 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
    • G10L19/0212—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 orthogonal transformation
    • 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
    • G10L25/21—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 power information
    • 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
    • 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
    • G10L19/26—Pre-filtering or post-filtering
    • 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

Definitions

  • the present invention relates to the field of coding/decoding and processing of audio frequency signals (such as speech, music or other signals) for their transmission or storage.
  • audio frequency signals such as speech, music or other signals
  • the invention relates to a method and a device for frequency band extension in a decoder or a processor performing audio frequency signal improvement.
  • the limitation of the coded band of the AMR-WB codec to 7kHz is essentially linked to the fact that the transmission frequency response of wideband terminals was approximated at the time of standardization (ETSI/3GPP then ITU- T) according to the frequency mask defined in the ITU-T P.341 standard and more precisely by using a filter called “P341” defined in the ITU-T G.191 standard which cuts frequencies above 7 kHz (this filter respects the mask defined in P.341).
  • the 3GPP AMR-WB speech code was standardized in 2001 primarily for circuit mode (CS) telephony applications over GSM (2G) and UMTS (3G). This same code was also standardized in 2003 at the ITU-T as recommendation G.722.2 "Wideband coding speech at around 16kbit/s using Adaptive Multi-Rate Wideband (AMR-WB)".
  • DTX Discontinuous Transmission
  • VAD Voice Activity Detection
  • CNG Noise Generation
  • FEC Frequency Insertion Descriptor
  • band extension in AMR-WB codec is quite rudimentary. Indeed, the high band (6.4-7 kHz) is generated by shaping white noise through a temporal envelope (applied in the form of gains per subframe) and frequency (by the application of a linear prediction synthesis filter or LPC for “Linear Predictive Coding”). This band extension technique is illustrated in figure 1 .
  • correction information is transmitted by the AMR-WB encoder and decoded (blocks 107, 108) in order to refine the estimated gain per subframe (4 bits every 5ms, i.e. 0.8 kbit/s) .
  • the ITU-T G.718 standard includes a so-called interoperable mode, for which the core coding is compatible with G.722.2 (AMR-WB) coding at 12.65 kbit/s; in addition, the G.718 decoder has the particularity of being able to decode an AMR-WB/G.722.2 binary stream at all possible bit rates of the AMR-WB code (from 6.6 to 23.85 kbit/s).
  • the synthesis of high frequencies by shaped white noise is a very limited model of the signal in the frequency band above 6.4 kHz.
  • the present invention improves the situation.
  • the invention proposes a method for extending the frequency band of an audio frequency signal during a decoding or improvement process comprising a step of obtaining the decoded signal in a first frequency band called low band, as claimed by claim 1.
  • the "band extension” will be taken in the broad sense and will include not only the case of the extension of a sub-band at high frequencies but also the case of replacing sub-bands set to zero (of the “noise filling” type in transform coding).
  • both taking into account tonal components and an ambient signal extracted from the signal resulting from the decoding of the low band makes it possible to carry out the band extension with a signal model adapted to the true nature of the signal unlike the use of artificial noise.
  • the quality of the band extension is thus improved, particularly for certain types of signals such as music signals.
  • the signal decoded in the low band includes a part corresponding to the sound environment which can be transposed to high frequency such that a mixing of the harmonic components and the existing atmosphere makes it possible to ensure a reconstructed high band coherent.
  • band extension in particular in an enhancement device performing an analysis of the audio signal to extract the parameters necessary for band extension.
  • the band extension is performed in the excitation domain and the decoded low-band signal is a decoded low-band excitation signal.
  • the advantage of this embodiment is that a transformation without windowing (or equivalently with an implicit rectangular window of the frame length) is possible in the excitation domain. In this case no artifacts (block effects) are then audible.
  • the decoded low-band signal undergoes a step of decomposition into sub-bands by transform or by filter bank, the extraction and combination steps then being carried out in the frequency domain or in sub-bands. .
  • the implementation of the band extension in the frequency domain makes it possible to obtain a fineness of frequency analysis which is not available with a temporal approach, and also makes it possible to have sufficient frequency resolution to detect the tonal components. .
  • this function includes resampling the signal by adding samples to the spectrum of this signal.
  • Other ways of extending the signal are however possible, for example by translation in sub-band processing.
  • the present invention also relates to a device for extending the frequency band of an audio frequency signal, the signal having been decoded in a first frequency band called low band, as claimed by claim 9.
  • This device has the same advantages as the method described above, which it implements.
  • the invention relates to a decoder comprising a device as described.
  • the invention relates to a storage medium, readable by a processor, integrated or not into the tape extension device, possibly removable, memorizing a computer program implementing a tape extension method as described above.
  • FIG. 3 illustrates an example of decoder, compatible with the AMR-WB/G.722.2 standard in which we find post-processing similar to that introduced in G.718 and described with reference to the figure 2 and an improved band extension according to the extension method of the invention, implemented by the band extension device illustrated by block 309.
  • AMR-WB decoding which operates with an output sampling frequency of 16 kHz
  • G.718 decoding which operates at 8 or 16 kHz
  • CELP decoding (LF for low frequencies) always works at the internal frequency of 12.8 kHz, as in AMR-WB and G.718, and the band extension (HF for high frequencies) which is the subject of the invention operates at a frequency of 16 kHz, the LF and HF syntheses are combined (block 312) at the frequency fs after adequate resampling (blocks 307 and 311).
  • the combination of the low and high bands can be done at 16 kHz, after having resampled the low band from 12.8 to 16 kHz, before resampling the combined signal at the frequency fs.
  • the post-processings applied to the excitation can be modified (for example, the phase dispersion can be improved) or these post-processings can be extended (for example, a reduction of inter-harmonic noise can be implemented), without affecting the nature of the band extension.
  • We do not describe here the case of low band decoding when the current frame is lost (bfi 1) which is informative in the 3GPP AMR-WB standard; in general, whether it is the AMR-WB decoder or a general decoder based on the source-filter model, it is typically a question of best estimating the LPC excitation and the coefficients of the LPC filter synthesis in order to reconstitute the lost signal while keeping the source-filter model.
  • the low band decoding described above assumes a current so-called “active” frame with a bit rate between 6.6 and 23.85 kbit/s.
  • active a current so-called “active” frame with a bit rate between 6.6 and 23.85 kbit/s.
  • certain frames can be coded as “inactive” and in this case you can either transmit a silence descriptor (on 35 bits) or not transmit anything.
  • SID frame of the AMR-WB encoder describes several parameters: ISF parameters averaged over 8 frames, average energy over 8 frames, "dithering flag" for the reconstruction of non-stationary noise.
  • This example of decoder operates in the excitation domain and therefore includes a step of decoding the low-band excitation signal.
  • the band extension device and the band extension method within the meaning of the invention also operate in a field other than the field of excitation and in particular with a direct signal decoded in low band or a signal weighted by a filter perceptual.
  • the decoder described makes it possible to extend the decoded low band (50-6400 Hz taking into account high-pass filtering at 50 Hz at the decoder, 0-6400 Hz in the general case ) to an extended band whose width varies, ranging approximately from 50-6900 Hz to 50-7700 Hz depending on the mode implemented in the current frame.
  • the excitation for high frequencies and generated in the frequency domain in a band from 5000 to 8000 Hz, to allow bandpass filtering of width 6000 to 6900 or 7700 Hz whose slope is not too steep in the upper rejected band.
  • the high band synthesis part is carried out in block 309 representing the band extension device according to the invention and which is detailed in Figure 5 in one embodiment.
  • a delay (block 310) is introduced to synchronize the outputs of blocks 306 and 309 and the high band synthesized at 16 kHz is resampled from 16 kHz to frequency fs (output of block 311).
  • fs 8 kHz, it is not necessary to apply blocks 309 to 311 because the signal band output from the decoder is limited to 0-4000 Hz.
  • the extension method of the invention implemented in block 309 according to the first embodiment preferentially does not introduce any additional delay compared to the low band reconstructed at 12.8 kHz; however, in variants of the invention (for example using a time/frequency transformation with overlap), a delay may be introduced.
  • the low and high bands are then combined (added) in block 312 and the synthesis obtained is post-processed by high-pass filtering at 50 Hz (IIR type) of order 2 whose coefficients depend on the frequency fs (block 313) and output post-processing with optional application of the "noise gate” in a manner similar to G.718 (block 314).
  • the band extension device according to the invention illustrated by block 309 according to the embodiment of the decoder of the Figure 5 , implements a band extension process (in the broad sense) now described with reference to the figure 4 .
  • This extension device can also be independent of the decoder and can implement the method described in figure 4 to carry out a band extension of an existing audio signal stored or transmitted to the device, with an analysis of the audio signal to extract for example an excitation and an LPC filter.
  • This device receives as input a decoded signal in a first frequency band called low band u(n) which can be in the excitation domain or in that of the signal.
  • a step of decomposition into sub-bands (E401b) by time-frequency transform or filter bank is applied to the low-band decoded signal to obtain the spectrum of the low-band decoded signal U(k) for an update implemented in the frequency domain.
  • This extension step can include both a resampling step and an extension step or simply a frequency translation or transposition step depending on the signal obtained at the input. Note that in variants not covered by the claims, step E401a may be carried out at the end of the processing described in Figure 4 ,, that is to say on the combined signal, this processing then being mainly carried out on the low band signal before extension, the result being equivalent.
  • a step E402 of extracting an ambient signal ( U HBA ( k )) and tonal components (y(k)) is carried out from the decoded low band signal ( U ( k )) or decoded and extended ( U HB 1 ( k )) .
  • ambience here as the residual signal which is obtained by removing the main (or dominant) harmonics (or tonal components) from the existing signal.
  • the high band In most wideband signals (sampled at 16 kHz), the high band (>6 kHz) contains ambient information that is generally similar to that present in the low band.
  • step E403 The tonal components and the ambient signal are then combined adaptively using energy level control factors in step E403 to obtain a so-called combined signal ( U HB 2 ( k )).
  • the extension step E401a can then be implemented if it has not already been carried out on the decoded low band signal.
  • the combination of these two types of signals makes it possible to obtain a combined signal with characteristics more suited to certain types of signals such as musical signals and richer in frequency content and in the extended frequency band corresponding to the entire band of frequency including the first and second frequency bands.
  • the band extension according to the method improves the quality for this type of signals compared to the extension described in the AMR-WB standard.
  • a synthesis step which corresponds to the analysis in 401b, is carried out in E404b to bring the signal back into the time domain.
  • a step of adjusting the energy level of the high band signal can be carried out in E404a, before and/or after the synthesis step, by application of a gain and/or by appropriate filtering. This step will be explained in more detail in the embodiment described in section Figure 5 for blocks 501 to 507.
  • the band extension device 500 is now described with reference to the Figure 5 illustrating both this device but also processing modules suitable for implementation in an interoperable type decoder with AMR-WB coding.
  • This device 500 implements the band extension method described previously with reference to the Figure 4 .
  • the processing block 510 receives a decoded low-band signal ( u ( n )).
  • the band extension uses the excitation decoded at 12.8 kHz (exc2 or u ( n )) at the output of block 302 of the Figure 3 .
  • This signal is decomposed into frequency sub-bands by the sub-band decomposition module 510 (which implements step E401b of the Figure 4 ) which generally carries out a transform or applies a bank of filters, to obtain a decomposition into sub-bands U(k) of the signal u(n).
  • the sub-band decomposition module 510 which implements step E401b of the Figure 4 .
  • a windowless transformation (or equivalently with an implicit rectangular window of the frame length) is possible when processing is performed in the excitation domain, not the signal domain. In this case no artifact (block effects) is audible, which constitutes an important advantage of this embodiment of the invention.
  • the DCT-IV transformation is implemented by FFT following the so-called “Evolved DCT (EDCT )” algorithm described in the article by DM Zhang, HT Li, A Low Complexity Transform - Evolved DCT, IEEE 14th International Conference on Computational Science and Engineering (CSE), Aug. 2011, pp. 144-149 , and implemented in ITU-T standards G.718 Annex B and G.729.1 Annex E.
  • the DCT-IV transformation can be replaced by other short-term time-frequency transformations of the same length and in the excitation domain or in the signal domain, such as an FFT (for "Fast Fourier Transform” in English ) or a DCT-II ( Discrete Cosine Transform - Type II).
  • FFT Fast Fourier Transform
  • DCT-II Discrete Cosine Transform - Type II
  • MDCT for "Modified Discrete Cosine Transform” in English
  • the delay T in block 310 of the Figure 3 will have to be adjusted (reduced) adequately according to the additional delay due to the analysis/synthesis by this transform.
  • the decomposition into sub-bands is carried out by the application of a bank of filters, for example of the real or complex PQMF (Pseudo-QMF) type.
  • a bank of filters for example of the real or complex PQMF (Pseudo-QMF) type.
  • PQMF Pseudo-QMF
  • the preferred embodiment in the invention can be applied by carrying out for example a transform of each sub-band and calculating the ambient signal in the domain of absolute values, the tonal components always being obtained by difference between the signal (in absolute value) and the ambient signal.
  • the complex modulus of the samples will replace the absolute value.
  • the invention will be applied in a system using two sub-bands, the low band being analyzed by transform or by filter bank.
  • Block 511 implements step E401a of the Figure 4 , that is to say the extension of the low band decoded signal.
  • the original spectrum is preserved, to be able to apply a progressive attenuation response of the high-pass filter in this frequency band and also to not introduce audible defects during the step of adding the low frequency synthesis to the high frequency synthesis.
  • the generation of the over-sampled extended spectrum is carried out in a frequency band ranging from 5 to 8 kHz therefore including a second frequency band (6.4-8kHz) greater than the first frequency band (0-6.4 kHz).
  • the extension of the decoded low band signal is carried out at least on the second frequency band but also on part of the first frequency band.
  • the 6000-8000 Hz band of U HB 1 ( k ) is defined here by copying the 4000-6000 Hz band of U(k) since the value of start_band is preferably fixed at 160.
  • the value of start_band can be made adaptive around the value of 160, without modifying the nature of the invention.
  • the details of adapting the start_band value are not described here because they go beyond the scope of the invention without changing its scope.
  • the high band (>6 kHz) contains ambient information that is naturally similar to that present in the low band.
  • the ambience here as the residual signal which is obtained by removing the main (or dominant) harmonics from the existing signal. Harmonicity level in the 6000-8000 Hz band is generally correlated with that of lower frequency bands.
  • This decoded and extended low band signal is supplied at the input of the extension device 500 and in particular at the input of the module 512.
  • the block 512 for extracting tonal components and an ambient signal implements the step E402 of the figure 4 in the frequency domain.
  • L 80 and represents the length of the spectrum and the index i from 0 to L-1 corresponds to the indices j + 240 from 240 to 319, i.e. the spectrum from 6 to 8 kHz.
  • This variant has the drawback of being more complex (in terms of number of calculations ) than a rolling average.
  • a non-uniform weighting could be applied to the averaged terms, or the median filtering could be replaced, for example, by other non-linear filters of the “ stack filters” type.
  • This calculation therefore involves implicit detection of tonal components.
  • the tonal parts are therefore implicitly detected using the intermediate term y(i) representing an adaptive threshold.
  • the detection condition being y(i) >0.
  • this ambient signal can be extracted from a low frequency signal or possibly another frequency band (or several frequency bands).
  • the detection of peaks or tonal components can be done differently.
  • This ambient signal could also be done on the decoded excitation but not extended, that is to say before the spectral extension or translation step, that is to say for example on a portion of the low frequency signal rather than directly on the high frequency signal.
  • a peak (or tonal component) is detected at a line of index i in the amplitude spectrum
  • if the following criterion is verified: U H.B. 1 i + 240 > U H.B. 1 i + 240 ⁇ 1 And U H.B. 1 i + 240 > U H.B. 1 i + 240 + 1 , for i 0,..., L - 1.
  • a sinusoidal model is applied in order to estimate the amplitude, frequency and possibly phase parameters of a tonal component associated with this peak.
  • the frequency estimation can typically use a 3-point parabolic interpolation in order to locate the maximum of the parabola approximating the 3 amplitude points
  • the transform domain used here (DCT-IV) does not make it possible to directly obtain the phase, we can in one embodiment neglect this term, but in variants we can apply a quadrature transform of the DST type to estimate a phase term.
  • the sinusoidal parameters (frequency, amplitude, and possibly phase) of each tonal component being estimated we then calculate the term y(i) as the sum of predefined prototypes (spectra) of pure sinusoids transformed in the DCT-IV domain (or other if another decomposition into sub-bands is used) according to the estimated sinusoidal parameters. Finally, we apply an absolute value to the terms y(i) to return to the domain of the amplitude spectrum in absolute values.
  • the absolute value of the spectral values will be replaced, for example the square of the spectral values, without changing the principle of the invention; in this case a square root will be necessary to return to the signal domain, which is more complex to achieve.
  • the combination module 513 performs a combination step by adaptive mixing of the ambient signal and the tonal components.
  • the factor ⁇ is > 1.
  • the tonal components, detected line by line by the condition y(i)> 0, are reduced by the factor ⁇ ; the average level is amplified by the factor 1/ ⁇ .
  • an energy level control factor is calculated based on the total energy of the decoded (or decoded and expanded) low-band signal and the tonal components.
  • ⁇ makes it possible to avoid overestimation of the energy.
  • ⁇ is calculated so as to keep the same level of ambient signal in relation to the energy of the tonal components in the consecutive bands of the signal.
  • E NOT 2 ⁇ 4 ⁇ k ⁇ NOT 80,159 U ′ 2 k
  • E NOT 4 ⁇ 6 ⁇ k ⁇ NOT 160,239 U ′ 2 k
  • E NOT 4 ⁇ 6 ⁇ k ⁇ NOT 240,319 U ′ 2 k
  • N( k 1 , k 2 ) is the set of indices k for which the index coefficient k is classified as being associated with the tonal components.
  • This set can for example be obtained by detecting local peaks in U' ( k ) satisfying
  • E NOT 4 ⁇ 6 max E NOT 4 ⁇ 6 E NOT 2 ⁇ 4
  • ⁇ E NOT 4 ⁇ 6 2 E NOT 2 ⁇ 4
  • ⁇ max ⁇ E NOT 6 ⁇ 8
  • max(.,.) is the function which gives the maximum of the two arguments.
  • the calculation of ⁇ may be replaced by other methods.
  • the linear regression could for example be estimated in a supervised manner by estimating the factor ⁇ by giving the original high band in a base d 'learning. It should be noted that the method of calculating ⁇ does not limit the nature of the invention.
  • ⁇ and ⁇ are possible within the framework of the invention.
  • the block 501 performs a double operation of applying the frequency response of the band-pass filter and de-emphasis (or de-emphasis) filtering in the frequency domain.
  • the de-emphasis filtering could be carried out in the time domain, after block 502 or even before block 510; however, in this case, the bandpass filtering carried out in block 501 can leave certain low frequency components of very low levels which are amplified by deemphasis, which can modify the decoded low band in a slightly perceptible manner. For this reason, we prefer here to carry out the de-emphasis in the frequency domain.
  • ⁇ k can be adjusted (for example for even frequencies).
  • the high-frequency signal is on the contrary de-emphasized so as to bring it back into a domain coherent with the low-frequency signal (0-6.4 kHz) which leaves block 305 of the Figure 3 . This is important for the estimation and subsequent adjustment of the energy of HF synthesis.
  • the de-emphasis can be carried out equivalently in the time domain after inverse DCT.
  • bandpass filtering is applied with two separate parts: one fixed high pass, the other adaptive low pass (depending on the bitrate).
  • This filtering is carried out in the frequency domain.
  • Table 1 Table 1 ⁇ /b> K gnp ( k ) K ghp ( k ) K ghp ( k ) k ghp ( k ) 0 0.001622428 14 0.114057967 28 0.403990611 42 0.776551214 1 0.004717458 15 0.128865425 29 0.430149896 43 0.800503267 2 0.008410494 16 0.144662643 30 0.456722014 44 0.823611104 3 0.012747280 17 0.161445005 31 0.483628433 45 0.845788355 4 0.017772424 18 0.179202219 32 0.510787115 46 0.866951597 5 0.023528982 19 0.197918220 33 0.538112915 47 0.887020781 6 0.030058032 20 0.217571104 34 0.565518011 48 0.9059
  • G hp ( k ) can be modified while maintaining progressive attenuation.
  • low-pass filtering with variable bandwidth, G lp ( k ) can be adjusted with different values or frequency support, without changing the principle of this filtering step.
  • bandpass filtering can be adapted by defining a single filtering step combining high-pass and low-pass filtering.
  • the bandpass filtering could be carried out equivalently in the time domain (as in block 112 of the figure 1 ) with different filter coefficients depending on the flow rate, after an inverse DCT step.
  • it is advantageous to carry out this step directly in the frequency domain because the filtering is carried out in the LPC excitation domain and therefore the problems of circular convolution and edge effects are very limited in this domain. .
  • block 502 performs the synthesis corresponding to the analysis carried out in block 510.
  • the signal sampled at 16 kHz is then optionally scaled by gains defined per subframe of 80 samples (block 504).
  • block 503 differs from that of block 101 of the figure 1 , because the energy at the current frame is taken into account in addition to that of the subframe. This allows us to have the ratio of the energy of each subframe compared to the energy of the frame. We therefore compare energy ratios (or relative energies) rather than absolute energies between low band and high band.
  • this scaling step makes it possible to maintain in the high band the energy ratio between the subframe and the frame in the same way as in the low band.
  • Blocks 505 and 506 are useful for adjusting the level of the LPC synthesis filter (block 507), here as a function of the tilt of the signal. Other methods of calculating the gain g HB 2 ( m ) are possible without changing the nature of the invention.
  • this filtering could be carried out in the same way as what is described for block 111 of the figure 1 of the AMR-WB decoder, however the order of the filter increases to 20 at a bit rate of 6.6, which does not significantly change the quality of the synthesized signal.
  • LPC synthesis filtering can be carried out in the frequency domain, after having calculated the frequency response of the filter implemented in block 507.
  • the coding of the low band (0-6.4 kHz) could be replaced by a CELP coder other than that used in AMR-WB, such as for example the CELP coder in G.718 at 8 kbps.
  • a CELP coder other than that used in AMR-WB, such as for example the CELP coder in G.718 at 8 kbps.
  • other wideband encoders or those operating at frequencies above 16 kHz, in which the low band coding operates at an internal frequency of 12.8 kHz could be used.
  • the invention can obviously be adapted to sampling frequencies other than 12.8 kHz, when a low frequency encoder operates at a sampling frequency lower than that of the original or reconstructed signal.
  • the excitation or the low band signal ( u ( n )) is re-sampled, for example by linear interpolation or cubic "spline", of 12.8 to 16 kHz before transformation (for example DCT-IV) of length 320.
  • This variant has the disadvantage of being more complex, because the transform (DCT-IV) of the excitation or signal is then calculated on a larger length and resampling is not performed in the transform domain.
  • FIG. 6 represents an example of hardware embodiment of a band extension device 600 according to the invention. This can be an integral part of an audio frequency signal decoder or of equipment receiving decoded or non-decoded audio frequency signals.
  • This type of device comprises a processor PROC cooperating with a memory block BM comprising a storage and/or working memory MEM.
  • Such a device comprises an input module E capable of receiving an audio signal decoded or extracted in a first frequency band called low band brought back into the frequency domain ( U ( k )). It comprises an output module S capable of transmitting the extension signal in a second frequency band ( U HB 2 ( k )) for example to a filter module 501 of the Figure 5 .
  • the memory block may advantageously comprise a computer program comprising code instructions for implementing the steps of the band extension method within the meaning of the invention, when these instructions are executed by the processor PROC, and in particular the steps extraction (E402) of tonal components and an ambient signal from a signal from the decoded low band signal ( U ( k )) , combination (E403) of the tonal components (y (k)) and the ambient signal ( U HBA ( k )) by adaptive mixing using energy level control factors to obtain an audio signal, called combined signal ( U HB 2 ( k )), extension (E401a) on at least a second frequency band greater than the first frequency band of the low-band decoded signal before the extraction step or of the combined signal after the combination step.
  • a computer program comprising code instructions for implementing the steps of the band extension method within the meaning of the invention, when these instructions are executed by the processor PROC, and in particular the steps extraction (E402) of tonal components and an ambient signal from a signal from the decoded low band
  • the description of the Figure 4 repeats the steps of an algorithm of such a computer program.
  • the computer program can also be stored on a memory medium readable by a reader of the device or downloadable into the memory space thereof.
  • the MEM memory generally records all the data necessary for implementing the process.
  • the device thus described can also include the low-band decoding functions and other processing functions described for example in Figure 5 And 3 in addition to the band extension functions according to the invention.

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US20180141361A1 (en) 2018-05-24
SI3330966T1 (sl) 2023-12-29
US20180304659A1 (en) 2018-10-25
KR20160119150A (ko) 2016-10-12
ZA201708368B (en) 2018-11-28
CN105960675A (zh) 2016-09-21
CN107993667B (zh) 2021-12-07
KR20180002910A (ko) 2018-01-08
RU2017144522A3 (ko) 2021-04-01
PL3330966T3 (pl) 2023-12-18
ZA201708366B (en) 2019-05-29
US10668760B2 (en) 2020-06-02
KR20180002907A (ko) 2018-01-08
RU2017144523A3 (ko) 2021-04-01
US10043525B2 (en) 2018-08-07
PT3103116T (pt) 2021-07-12
JP2019168708A (ja) 2019-10-03
RU2016136008A3 (ko) 2018-09-13
US20170169831A1 (en) 2017-06-15
US20200338917A1 (en) 2020-10-29
PL3103116T3 (pl) 2021-11-22
HRP20211187T1 (hr) 2021-10-29
LT3103116T (lt) 2021-07-26
US10730329B2 (en) 2020-08-04

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