EP3330966B1 - Verbesserte ausdehnung eines frequenzbands in einem dekodierer von audiofrequenzsignalen - Google Patents

Verbesserte ausdehnung eines frequenzbands in einem dekodierer von audiofrequenzsignalen Download PDF

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EP3330966B1
EP3330966B1 EP17206563.3A EP17206563A EP3330966B1 EP 3330966 B1 EP3330966 B1 EP 3330966B1 EP 17206563 A EP17206563 A EP 17206563A EP 3330966 B1 EP3330966 B1 EP 3330966B1
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signal
band
frequency
khz
tonal components
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EP3330966A1 (de
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Magdalena KANIEWSKA
Stéphane RAGOT
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Koninklijke Philips NV
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Koninklijke Philips NV
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Priority to RS20230844A priority patent/RS64614B1/sr
Priority to SI201531958T priority patent/SI3330966T1/sl
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    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10LSPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L21/00Speech 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
    • G10L21/02Speech enhancement, e.g. noise reduction or echo cancellation
    • G10L21/038Speech enhancement, e.g. noise reduction or echo cancellation using band spreading techniques
    • BPERFORMING OPERATIONS; TRANSPORTING
    • B41PRINTING; LINING MACHINES; TYPEWRITERS; STAMPS
    • B41KSTAMPS; STAMPING OR NUMBERING APPARATUS OR DEVICES
    • B41K3/00Apparatus for stamping articles having integral means for supporting the articles to be stamped
    • B41K3/54Inking devices
    • B41K3/56Inking devices using inking pads
    • BPERFORMING OPERATIONS; TRANSPORTING
    • B41PRINTING; LINING MACHINES; TYPEWRITERS; STAMPS
    • B41KSTAMPS; STAMPING OR NUMBERING APPARATUS OR DEVICES
    • B41K1/00Portable hand-operated devices without means for supporting or locating the articles to be stamped, i.e. hand stamps; Inking devices or other accessories therefor
    • B41K1/02Portable 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
    • B41K1/04Portable 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
    • BPERFORMING OPERATIONS; TRANSPORTING
    • B41PRINTING; LINING MACHINES; TYPEWRITERS; STAMPS
    • B41KSTAMPS; STAMPING OR NUMBERING APPARATUS OR DEVICES
    • B41K1/00Portable hand-operated devices without means for supporting or locating the articles to be stamped, i.e. hand stamps; Inking devices or other accessories therefor
    • B41K1/08Portable 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
    • B41K1/10Portable 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
    • BPERFORMING OPERATIONS; TRANSPORTING
    • B41PRINTING; LINING MACHINES; TYPEWRITERS; STAMPS
    • B41KSTAMPS; STAMPING OR NUMBERING APPARATUS OR DEVICES
    • B41K1/00Portable hand-operated devices without means for supporting or locating the articles to be stamped, i.e. hand stamps; Inking devices or other accessories therefor
    • B41K1/08Portable 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
    • B41K1/12Portable 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
    • BPERFORMING OPERATIONS; TRANSPORTING
    • B41PRINTING; LINING MACHINES; TYPEWRITERS; STAMPS
    • B41KSTAMPS; STAMPING OR NUMBERING APPARATUS OR DEVICES
    • B41K1/00Portable hand-operated devices without means for supporting or locating the articles to be stamped, i.e. hand stamps; Inking devices or other accessories therefor
    • B41K1/36Details
    • B41K1/38Inking devices; Stamping surfaces
    • BPERFORMING OPERATIONS; TRANSPORTING
    • B41PRINTING; LINING MACHINES; TYPEWRITERS; STAMPS
    • B41KSTAMPS; STAMPING OR NUMBERING APPARATUS OR DEVICES
    • B41K1/00Portable hand-operated devices without means for supporting or locating the articles to be stamped, i.e. hand stamps; Inking devices or other accessories therefor
    • B41K1/36Details
    • B41K1/38Inking devices; Stamping surfaces
    • B41K1/40Inking devices operated by stamping movement
    • BPERFORMING OPERATIONS; TRANSPORTING
    • B41PRINTING; LINING MACHINES; TYPEWRITERS; STAMPS
    • B41KSTAMPS; STAMPING OR NUMBERING APPARATUS OR DEVICES
    • B41K1/00Portable hand-operated devices without means for supporting or locating the articles to be stamped, i.e. hand stamps; Inking devices or other accessories therefor
    • B41K1/36Details
    • B41K1/38Inking devices; Stamping surfaces
    • B41K1/40Inking devices operated by stamping movement
    • B41K1/42Inking devices operated by stamping movement with pads or rollers movable for inking
    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10LSPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00Speech 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
    • G10L19/02Speech 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
    • G10L19/0204Speech 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
    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10LSPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00Speech 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
    • G10L19/02Speech 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
    • G10L19/0212Speech 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
    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10LSPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L25/00Speech or voice analysis techniques not restricted to a single one of groups G10L15/00 - G10L21/00
    • G10L25/03Speech or voice analysis techniques not restricted to a single one of groups G10L15/00 - G10L21/00 characterised by the type of extracted parameters
    • G10L25/21Speech 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
    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10LSPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00Speech 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
    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10LSPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00Speech 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
    • G10L19/04Speech 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
    • G10L19/26Pre-filtering or post-filtering
    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10LSPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L21/00Speech 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
    • G10L21/02Speech enhancement, e.g. noise reduction or echo cancellation

Definitions

  • the present invention relates to the field of the coding/decoding and processing of audio frequency signals (such as speech, music or other signals) for their transmission or their 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 an audio frequency signal enhancement.
  • 3GPP AMR-WB for "Adaptive Multi-Rate Wideband" which operates at an input/output frequency of 16 kHz and in which the signal is divided into two sub-bands, the low band (0-6.4 kHz) which is sampled at 12.8 kHz and coded by CELP model and the high band (6.4-7 kHz) which is reconstructed para metric by “Bandwidth Extension ” (or BWE) with or without additional information depending on the mode of the current frame.
  • codec coder and decoder
  • 3GPP AMR-WB for "Adaptive Multi-Rate Wideband”
  • the 3GPP AMR-WB speech codec was standardized in 2001 primarily for circuit mode (CS) telephony applications on GSM (2G) and UMTS (3G). This same codec 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)".
  • the principle of band extension in the AMR-WB codec is quite rudimentary. Indeed, the high band (6.4-7 kHz) is generated by shaping a white noise by means of 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 coder and decoded (blocks 107, 108) in order to refine the estimated gain per subframe (4 bits every 5 ms, ie 0.8 kbit/s).
  • s HB ( n ) is finally processed by a band-pass filter (block 112) of the FIR ("Finite Impulse Response") type, to keep only the 6-7 kHz band; at 23.85 kbit/s, a low-pass filter also of the FIR type (block 113) is added to the processing to further attenuate the frequencies above 7 kHz.
  • the high frequency (HF) synthesis is finally added (block 130) to the low frequency (LF) synthesis obtained with blocks 120 to 123 and resampled at 16 kHz (block 123).
  • the HF synthesis is rather included in the 6-7 kHz band before addition with the LF synthesis.
  • the AMR-WB decoding algorithm was improved in part with the development of the ITU-T G.718 scalable codec which was standardized in 2008.
  • the ITU-T G.718 standard includes a so-called interoperable mode, for which the core coding is compatible with the G.722.2 (AMR-WB) coding at 12.65 kbit/s; moreover, the G.718 decoder has the particularity of being able to decode an AMR-WB/G.722.2 binary train at all possible rates of the AMR-WB codec (from 6.6 to 23.85 kbit/s).
  • the interoperable G.718 decoder in low-delay mode (G.718-LD) is shown in Fig. figure 2 .
  • Fig. figure 2 The band expansion (described for example in clause 7.13.1 of recommendation G.718, block 206) is identical to that of the AMR-WB decoder, except that the 6-7 kHz bandpass filter and the 1/A HB (z) synthesis filter (blocks 111 and 112) are in reverse order.
  • the 4 bits transmitted by subframes by the AMR-WB coder are not used in the interoperable G.718 decoder; the synthesis of high frequencies (HF) at 23.85 kbit/s is therefore identical to 23.05 kbit/s, which avoids the known quality problem of AMR-WB decoding at 23.85 kbit/s.
  • HF high frequencies
  • the low-pass filter at 7 kHz block 113 is not used, and the specific decoding of the mode at 23.85 kbit/s is omitted (blocks 107 to 109).
  • a post-processing of the synthesis at 16 kHz is implemented in G.718 by " noise gate " in block 208 (to "improve” the quality of silences by level reduction), high-pass filtering (block 209), low-frequency post-filter (called “ bass posfilier ”) in block 210 attenuating inter-harmonic noise at low frequencies and a conversion to 16-bit integers with saturation control (with gain control or AGC) in block 211.
  • the synthesis of high frequencies by shaped white noise is a very limited model of the signal in the band of frequencies above 6.4 kHz.
  • the present invention improves the situation.
  • the invention proposes for this purpose 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 so-called low band frequency band.
  • the method is such that it includes the steps of claim 1.
  • 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 a replacement of sub-bands set to zero (of the "noise filling" type in transform coding).
  • the signal decoded in the low band comprises a part corresponding to the sound environment which can be transposed at high frequency in such a way that a mixing of the harmonic components and of the existing atmosphere makes it possible to ensure a coherent reconstructed high band.
  • the band expansion 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 artefact (block effects) is then audible.
  • This embodiment allows precise detection of the tonal components.
  • an energy level control factor used for the adaptive mixing is calculated based on the total energy of the decoded or decoded and extended low-band signal and the tonal components.
  • 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 performed in the frequency domain or in sub-bands.
  • the implementation of the band extension in the frequency domain makes it possible to obtain a precision of frequency analysis which is not available with a temporal approach, and also makes it possible to have a frequency resolution sufficient to detect the tonal components.
  • this function includes a resampling of 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.
  • This device has the same advantages as the method described previously, which it implements.
  • the invention relates to a decoder comprising a device as described.
  • It relates to a computer program comprising code instructions for implementing the steps of the band extension method as described, when these instructions are executed by a processor.
  • the invention relates to a storage medium, readable by a processor, integrated or not in the tape extension device, optionally removable, storing a computer program implementing a tape extension method as described previously.
  • FIG. 3 illustrates an example of a decoder, compatible with the AMR-WB/G.722.2 standard in which there is a post-processing similar to that introduced in G.718 and described with reference to figure 2 and an improved tape extension according to the extension method of the invention, implemented by the tape extension device illustrated by block 309.
  • the CELP decoding (LF for low frequencies) always operates 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 the frequency of 16 kHz, the LF and HF synthesis are combined (block 312) at the frequency fs after adequate resampling (blocks 307 and 311).
  • the combination of the low and high bands could 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 the inter-harmonic noise can be implemented), without affecting the nature of the band extension.
  • the decoding of the low band 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.
  • some frames can be coded as "inactive” and in this case you can either transmit a silence descriptor (on 35 bits) or transmit nothing.
  • SID frame of the AMR-WB coder describes several parameters: ISF parameters averaged over 8 frames, average energy over 8 frames, "dithering flag" for the reconstruction of non-stationary noise.
  • the same decoding model is found as for an active frame, with reconstruction of the excitation and of an LPC filter for the current frame, which makes it possible to apply the invention even to inactive frames.
  • the same observation applies for the decoding of “lost frames” (or FEC, PLC) in which the LPC model is applied.
  • This exemplary decoder operates in the excitation domain and therefore includes a step for 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 domain different from the domain of excitation and in particular with a direct signal decoded in low band or a signal weighted by a perceptual filter.
  • the decoder described makes it possible to extend the decoded low band (50-6400 Hz taking into account the 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 is generated in the frequency domain in a band from 5000 to 8000 Hz, to allow a 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 the 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 synthesized 16 kHz high band is resampled by 16 kHz at frequency fs (output of block 311).
  • the extension method of the invention implemented in block 309 according to the first embodiment preferentially does not introduce any additional delay with respect to the reconstructed low band 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 (of 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 extender according to the invention illustrated by block 309 according to the decoder embodiment of the figure 5 , implements a band extension method (in the broad sense) described now 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 therefrom for example an excitation and an LPC filter.
  • This device receives as input a decoded signal in a first frequency band called the low band u ( n ) which can be in the field of excitation or in that of the signal.
  • a step of decomposition into sub-bands (E401b) by time-frequency transform or bank of filters is applied to the decoded low-band signal to obtain the spectrum of the decoded low-band signal U ( k ) for implementation in the frequency domain.
  • a step E401a of extending the low band decoded signal into a second frequency band higher than the first frequency band, to obtain an extended low band decoded signal U HB 1 ( k ), can be performed on this low band decoded signal before or after the analysis step (decomposition into sub-bands).
  • This extension step can comprise both a resampling step and an extension step or simply a frequency translation or transposition step as a function of the signal obtained as input.
  • step E401a may be performed 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 performed from the decoded low band signal ( U ( k )) or decoded and extended ( U HB 1 ( k )) .
  • Ambiance is defined 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 performed 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 frequency band including the first and the second frequency band.
  • Band extension according to the method improves the quality for this type of signal compared to the extension described in the AMR-WB standard.
  • a synthesis step which corresponds to the analysis at 401b, is carried out at E404b to bring the signal back into the time domain.
  • a step of adjusting the energy level of the high band signal can be performed in E404a, before and/or after the synthesis step, by applying a gain and/or by suitable filtering. This step will be explained in more detail in the embodiment described in figure 5 for blocks 501 to 507.
  • tape expander 500 is now described with reference to figure 5 illustrating both this device but also processing modules suitable for implementation in a decoder of the interoperable type with AMR-WB coding.
  • This device 500 implements the band extension method previously described with reference to the figure 4 .
  • processing block 510 receives a decoded low band signal ( u ( n )).
  • the band extension uses the decoded excitation at 12.8 kHz (exc2 or u ( n ) ) at the output of block 302 of the picture 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 performs a transform or applies a bank of filters, to obtain a decomposition into sub-bands U(k) of the signal u(n).
  • a windowless transformation (or equivalently with an implicit rectangular window of the frame length) is possible when the processing is done in the excitation domain, not the signal domain. In this case, no artefact (block effects) is audible, which constitutes an important advantage of this embodiment of the invention.
  • the DCT-IV transformation is implemented by FFT according to 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, p. 144-149 , and implemented in ITU-T G.718 Annex B and G.729.1 Annex E.
  • EDCT Evolved DCT
  • the DCT-IV transformation may 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” ) or a DCT-II ( Discrete Cosine Transform - Type II).
  • the DCT-IV on the frame can be replaced by a transformation with overlap-addition and windowing of length greater than the length of the current frame, for example by using an MDCT (for “ Modified Discrete Cosine Tranform ”).
  • MDCT Modified Discrete Cosine Tranform
  • the decomposition into sub-bands is performed by applying a bank of filters, for example of real or complex PQMF (Pseudo-QMF) type.
  • a bank of filters for example of 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 by 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 bank of filters.
  • 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 kept, in order to be able to apply a progressive attenuation response of the high-pass filter in this frequency band and also so as not to introduce audible defects during the step of adding the low-frequency synthesis to the high-frequency synthesis.
  • the generation of the oversampled extended spectrum is carried out in a frequency band ranging from 5 to 8 kHz, therefore including a second frequency band (6.4-8 kHz) higher than the first frequency band (0-6.4 kHz).
  • the extension of the decoded low band signal takes place 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 here defined by copying the 4000-6000 Hz band of U(k) since the value of start_band is preferentially fixed at 160.
  • start_band could be made adaptive around the value of 160, without modifying the nature of the invention.
  • the details of the adaptation of the start_band value are not described here because they exceed the scope of the invention without changing the scope thereof.
  • the high band In most wideband signals (sampled at 16 kHz), the high band (>6 kHz) contains ambient information that is naturally similar to that present in the low band. Ambiance is defined here as the residual signal which is obtained by removing the main (or dominant) harmonics from the existing signal. level of harmonicity 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 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, ie the spectrum from 6 to 8 kHz.
  • a non-uniform weighting may be applied to the averaged terms, or the median filtering may be replaced, for example, by other nonlinear filters of the “ stack filters ” type.
  • This calculation therefore involves an implicit detection of the 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).
  • 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 HB 1 I + 240 > U HB 1 I + 240 ⁇ 1 And U HB 1 I + 240 > U HB 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 to locate the maximum of the parabola approximating the 3 amplitude points
  • DCT-IV transform domain used here
  • the term y( i ) is then calculated 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, an absolute value is applied to the terms y( i ) to reduce 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 combining module 513 performs a combining 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 extended) low-band signal and the tonal components.
  • avoids an overestimation of the energy.
  • is calculated so as to keep the same ambient signal level with respect to the energy of the tonal components in the consecutive bands of the signal.
  • E NOT 2 ⁇ 4 ⁇ k ⁇ NOT 80.159
  • 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 coefficient of index k is classified as being associated with the tonal components.
  • This set can be for example obtained by detecting the local peaks in U' ( k ) verifying
  • the calculation of ⁇ could be replaced by other methods.
  • the linear regression could for example be estimated in a supervised way by estimating the factor ⁇ by giving the original high band in a learning base. It will be noted that the mode of calculation of ⁇ does not limit the nature of the invention.
  • ⁇ and ⁇ are possible within the framework of the invention.
  • the block 501 At the output of the band extender 500, the block 501, in a particular embodiment, optionally performs a double operation of applying bandpass filter frequency response and de-emphasis filtering (or de-emphasis) in the frequency domain.
  • the de-emphasis filtering could be performed in the time domain, after block 502 or even before block 510; however, in this case, the band-pass filtering performed in block 501 may leave some low frequency components of very low levels which are amplified by de-emphasis, which may change the decoded low band slightly perceptibly. For this reason, it is preferred here to carry out the de-emphasis in the frequency domain.
  • ⁇ k can be adjusted (for example for even frequencies).
  • the HF synthesis is not de-emphasized.
  • the high-frequency signal is on the contrary de-emphasized so as to bring it back into a coherent domain with the low-frequency signal (0-6.4 kHz) which comes out of block 305 of the picture 3 . This is important for estimating and later adjusting the energy of HF synthesis.
  • the de-emphasis could be carried out in an equivalent manner in the time domain after inverse DCT.
  • band-pass filtering is applied with two separate parts: one fixed high-pass, the other adaptive low-pass (depending on the bit rate).
  • This filtering is performed in the frequency domain.
  • G hp ( k ), k 0.55, is given for example in Table 1 below.
  • G hp ( k ) can be modified while keeping a progressive attenuation.
  • the low-pass filtering with variable bandwidth, G lp ( k ) can be adjusted with different values or a frequency support, without changing the principle of this filtering step.
  • the band-pass filtering can be adapted by defining a single filtering step combining high-pass and low-pass filtering.
  • the band-pass filtering could be carried out in an equivalent way in the time domain (as in block 112 of the figure 1 ) with different filter coefficients depending on the bit rate, after an inverse DCT step.
  • it is advantageous to carry out this step directly in the frequency domain since the filtering is carried out in the domain of the LPC excitation and therefore the problems of circular convolution and of edge effects are very limited in this domain.
  • block 502 performs the synthesis corresponding to the analysis performed in block 510.
  • the 16 kHz sampled signal is then optionally scaled by defined gains per 80-sample subframe (block 504).
  • block 503 differs from that of block 101 of the figure 1 , because the energy at the level of the current frame is taken into account in addition to that of the subframe. This makes it possible to have the ratio of the energy of each subframe compared to the energy of the frame. Energy ratios (or relative energies) are therefore compared rather than the absolute energies between low band and high band.
  • this scaling step makes it possible to preserve 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 according to 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 changes to 20 at the rate of 6.6, which does not significantly change the quality of the synthesized signal.
  • the LPC synthesis filtering can be performed 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 kbit/s.
  • a CELP coder other than that used in AMR-WB, such as for example the CELP coder in G.718 at 8 kbit/s.
  • other wideband coders or coders operating at frequencies above 16 kHz, in which the low band coder 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 coder operates at a sampling frequency lower than that of the original or reconstructed signal.
  • the excitation or the low band signal ( u ( n )) is resampled, for example by linear interpolation or cubic "spline", from 12.8 to 16 kHz before transformation (for example DCT-IV) of length 320.
  • This variant has the defect of being more complex, because the transform (DCT-IV) of the excitation or of the signal is then calculated over a greater length and the resampling is not carried out in the domain of the transform.
  • FIG. 6 shows an exemplary hardware embodiment of a band extender device 600 according to the invention. This may be an integral part of an audio frequency signal decoder or of equipment receiving decoded or undecoded 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 the low band brought back into the frequency domain ( U ( k )). It comprises an output module S able to transmit 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 can 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 of extracting (E402) tonal components and of an ambient signal from a signal originating from the decoded low-band signal ( U ( k )), of combining (E403) 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 )), of extension (E401a) over at least a second frequency band higher than the first frequency band of the decoded low band signal before the extraction step or of the combined signal after the combining 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 of extracting (E402) tonal components and of an ambient signal from a signal
  • 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 that can be read by a reader of the device or that can be downloaded into the memory space of the latter.
  • the memory MEM generally records all the data necessary for the implementation of the method.
  • the device thus described may also comprise 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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Claims (9)

  1. Verfahren zur Frequenzbanderweiterung eines Audiofrequenzsignals bei einem Decodier- oder Verbesserungsprozess, das einen Schritt zum Erhalten des decodierten Signals in einem Frequenzband, als Tiefband bezeichnet, umfasst, wobei das Verfahren dadurch gekennzeichnet ist, dass es die folgenden Schritte umfasst:
    - Erweiterung (E401a) auf mindestens ein zweites Frequenzband, das höher ist als das erste Frequenzband des decodierten Tiefbandsignals, um ein erweitertes decodiertes Tiefbandsignal U HB1(k), zu bilden, wobei k die Abtastwerte darstellt, die das Spektrum UHB1(k) decken;
    - Extraktion (E402) tonaler Komponenten und eines Umgebungssignals ausgehend von dem Signal, das aus dem erweiterten decodierten Tiefbandsignal stammt;
    - Kombination (E403) der tonalen Komponenten und des Umgebungssignals durch adaptives Mischen, das Energieniveau-Steuerfaktoren verwendet, um ein kombiniertes Signal zu erhalten;
    - Synthese (E404b) eines Audiosignals, um ein Signal, das aus dem kombinierten Signal stammt, in den Zeitbereich zurückzuführen;
    und wobei der Extraktionsschritt (E402) der tonalen Komponenten und des Umgebungssignals die folgenden Operationen umfasst:
    (a) Berechnung der Gesamtenergie des erweiterten decodierten Tiefbandsignals;
    (b) Berechnung der Umgebung in Absolutwert, der dem mittleren Pegel des linienweisen Spektrums entspricht, und Berechnung der Energie der dominanten tonalen Teile in dem Hochfrequenzspektrum.
  2. Verfahren nach Anspruch 1, wobei die Operation (a) zur Berechnung der Gesamtenergie des erweiterten decodierten Tiefbandsignals die Berechnung umfasst von ener HB = k = 240 319 U HB 1 k 2 + ε
    Figure imgb0051
    wobei ε=0,1.
  3. Verfahren nach Anspruch 1 oder 2, wobei der mittlere Pegel des linienweisen Spektrums durch die folgende Gleichung erhalten wird: lev i = 1 fn i fb i + 1 i = fb i fn i U HB 1 j + 240
    Figure imgb0052
    wobei fb(i) = 0 und fn(i)= i+7 für i=0,...,6
    fb(i)= i-7 und fn(i)= i+7 für i=7,...,L-8
    fb(i)= i-7 und fn(i)= L-l für i=L-7,. L-1 ,
    wobei L die Länge des Spektrums ist.
  4. Verfahren nach Anspruch 1, 2 oder 3, wobei die Berechnung der Energie der dominanten tonalen Komponenten die Berechnung des Restsignals umfasst: y i = U HB 1 i + 240 0 lev i , i = 0 , L 1 .
    Figure imgb0053
  5. Verfahren nach Anspruch 4, das einen Erfassungsschritt der tonalen Komponenten auf der Grundlage einer Erfassungsbedingung auf dem Restsignal y(i) umfasst.
  6. Verfahren nach Anspruch 5, wobei die Erfassungsbedingung die Folgende ist y(i)>0.
  7. Verfahren nach Anspruch 6, wobei die Energie der dominanten tonalen Komponenten definiert wird von ener tonal = i = 0 7 y i > 0 y i 2
    Figure imgb0054
  8. Frequenzbanderweiterungsvorrichtung eines Audiofrequenzsignals, wobei das Signal in einem ersten Frequenzband, als Tiefband bezeichnet, decodiert wurde, wobei die Vorrichtung dadurch gekennzeichnet ist, dass sie Folgendes umfasst:
    - ein Modul (511) zum Erweitern auf mindestens ein zweites Frequenzband, das höher ist als das erste Frequenzband, das auf dem decodierten Tiefbandsignal umgesetzt wird, um ein erweitertes decodiertes Tiefbandsignal UHB1 (k) zu bilden, wobei k die Abtastwerte darstellt, die das Spektrum UHB1(k) decken;
    - ein Modul (512) zur Extraktion tonaler Komponenten und eines Umgebungssignals, ausgehend von einem Signal, das aus dem erweiterten decodierten Tiefbandsignal stammt;
    - ein Modul (513) zu kombinieren der tonalen Komponenten und des Umgebungssignals durch adaptives Mischen, das Energieniveau-Steuerfaktoren verwendet, um ein kombiniertes Audiosignal zu erhalten;
    - ein Modul (502) zur Synthese eines Audiosignals, um das Signal, das aus dem kombinierten Signal stammt, in den Zeitbereich zurückzuführen;
    und wobei das Modul (512) zur Extraktion der tonalen Komponenten und des Umgebungssignals dazu angepasst ist, die folgenden Operationen durchzuführen:
    (a) Berechnung der Gesamtenergie des erweiterten decodierten Tiefbandsignals;
    (b) Berechnung der Umgebung in Absolutwert, der dem mittleren Pegel des linienweisen Spektrums entspricht, und Berechnung der Energie der dominanten tonalen Teile in dem Hochfrequenzspektrum.
  9. Audiofrequenz-Signaldecoder, dadurch gekennzeichnet, dass er eine FrequenzbandErweiterungsvorrichtung nach Anspruch 8 umfasst.
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US10668760B2 (en) 2020-06-02
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RU2016136008A (ru) 2018-03-13
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CN105960675A (zh) 2016-09-21
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