EP3103116B1 - Verbesserte frequenzbanderweiterung in einem audiosignaldecodierer - Google Patents

Verbesserte frequenzbanderweiterung in einem audiosignaldecodierer Download PDF

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EP3103116B1
EP3103116B1 EP15705687.0A EP15705687A EP3103116B1 EP 3103116 B1 EP3103116 B1 EP 3103116B1 EP 15705687 A EP15705687 A EP 15705687A EP 3103116 B1 EP3103116 B1 EP 3103116B1
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Prior art keywords
signal
band
decoded
frequency
extension
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English (en)
French (fr)
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EP3103116A1 (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 RS20210945A priority Critical patent/RS62160B1/sr
Priority to DK17206563.3T priority patent/DK3330966T3/da
Priority to EP17206567.4A priority patent/EP3330967B1/de
Priority to PL15705687T priority patent/PL3103116T3/pl
Priority to EP17206563.3A priority patent/EP3330966B1/de
Application filed by Koninklijke Philips NV filed Critical Koninklijke Philips NV
Priority to EP17206569.0A priority patent/EP3327722B1/de
Priority to SI201531646T priority patent/SI3103116T1/sl
Publication of EP3103116A1 publication Critical patent/EP3103116A1/de
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    • 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
    • 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
    • G10L25/21—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 their storage.
  • audio-frequency signals such as speech, music or other signals
  • the invention relates to a method and a frequency band extension device in a decoder or a processor performing an audio-frequency signal improvement.
  • Classical coding methods for conversational applications are generally classified into waveform coding (PCM for "Pulse Modulation and Coding", ADPCM for “Pulse Modulation and Adaptive Differential Coding", transform coding, etc.) , parametric coding (LPC for "Linear Predictive Coding” in English, sinusoidal coding 7) and hybrid parametric coding with a quantization of the parameters by "analysis by synthesis” including CELP coding (for "Code Excited Linear Prediction” in English) is the best known example.
  • 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 the wideband terminals was approximated at the time of standardization (ETSI / 3GPP then UIT- T) according to the frequency mask defined in standard ITU-T P.341 and more precisely by using a so-called "P341" filter defined in standard ITU-T G.191 which cuts frequencies above 7 kHz (this filter respects the mask defined in P.341).
  • the 3GPP AMR-WB speech codec was standardized in 2001 primarily for circuit mode (CS) telephony applications over 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 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 tape 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 AMR-WB decoding algorithm was improved in part with the development of the scalable ITU-T G.718 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 bit stream at all the possible bit 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 illustrated on figure 2 .
  • the band extension (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 synthesis 1 / AHB (z) (blocks 111 and 112) are in reverse order.
  • the 4 bits transmitted by sub-frames by the AMR-WB encoder 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 problem of quality of AMR-WB decoding at 23.85 kbit / s.
  • HF high frequencies
  • the 7 kHz low-pass filter (block 113) is not used, and the specific decoding of the 23.85 kbit / s mode is omitted (blocks 107 to 109).
  • a 16 kHz synthesis post-processing (see clause 7.14 of G.718) is implemented in G.718 by "noise gate” in block 208 (to “improve” the quality of the silences by reducing the level) , high-pass filtering (block 209), low-frequency post-filter (called “ bass posfilter ”) in block 210 attenuating inter-harmonic noise at low frequencies and conversion to 16-bit integers with saturation control (with control gain or AGC) in block 211.
  • band extension in the AMR-WB and / or G.718 codecs is still limited in several aspects.
  • 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.
  • ABET Audio Bandwidth Extension Toolkit
  • ASR Accurate Spectral Replacement
  • FSSM Fractal Self Similarity Model
  • MBTAC Multi-Band Temporal Amplitude Amplitude Coding. Coding
  • 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 the low band.
  • the method is as it comprises 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 placed. to zero (of the “noise filling” type in transform coding).
  • both the taking into account of tonal components and of a surround 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 sound. signal unlike the use of artificial noise.
  • the quality of the band extension is thus improved and in particular for certain types of signals such as music signals.
  • the decoded signal in the low band includes a part corresponding to the sound environment which can be transposed to high frequency so that a mixing of the harmonic components and the existing environment makes it possible to ensure a reconstructed high band. consistent.
  • the band extension is performed in the field of excitation 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 length of the frame) is possible in the field of the excitation. In this case no artifact (block effects) is then audible.
  • this control factor allows the combining step to adapt to the characteristics of the signal to optimize the relative proportion of surround signal in the mixture.
  • the energy level is thus controlled so as to avoid audible artifacts.
  • the decoded low-band signal undergoes a step of decomposition into sub-bands by transform or by bank of filters, 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 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 a sub-band processing.
  • the present invention also relates to a device for extending the frequency band of an audiofrequency signal, the signal having been decoded in a first frequency band called the low band.
  • the device is such that it comprises the characteristics of claim 6.
  • 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, storing a computer program implementing a tape extension method as described above.
  • the figure 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.
  • CELP decoding (BF 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 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 .
  • This example of a decoder operates in the field of excitation and therefore comprises 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 different from 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 allows the decoded low band to be extended (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 from approximately 50-6900 Hz to 50-7700 Hz depending on the mode implemented in the current frame.
  • the excitation for the high frequencies is generated in the frequency domain in a band from 5000 to 8000 Hz, to allow band-pass filtering of width 6000 to 6900 or 7700 Hz, the slope of which 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 at 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 the 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 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, the coefficients of which depend on the frequency fs (block 313) and output post-processing with optional application of the "noise gate” similar to G.718 (block 314).
  • the band extension device according to the invention illustrated by the block 309 according to the embodiment of the decoder of the figure 5 , implements a band extension method (in the broad sense) described now with reference to figure 4 .
  • This extension device can also be independent of the decoder and can implement the method described in figure 4 to perform 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 the low band u ( n ) which may 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 a setting. implemented in the frequency domain.
  • a step E401a of extending the low-band decoded signal into a second frequency band greater than the first frequency band, to obtain an extended low-band decoded signal U HB 1 ( k ), can be performed on this decoded low-band signal before or after the analysis step (decomposition into sub-bands).
  • This extension step can include both a resampling step and an extension step or simply a frequency translation or transposition step as a function of the signal obtained at the input. It will be noted that in variants, step E401a could 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 a surround signal ( U HBA ( k )) and tonal components (y (k)) is performed from the decoded ( U ( k )) or decoded and extended ( U HB 1 ( k )) .
  • the ambience 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 (> 6 kHz) contains surround information which is generally similar to that present in the low band.
  • step E403 The tonal components and the surround signal are then adaptively combined using energy level control factors in step E403 to obtain a so-called combined signal ( U HB2 ( 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. frequency including the first and the second frequency band.
  • 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 performed 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 applying a gain and / or by suitable filtering. This step will be explained in more detail in the embodiment described in section figure 5 for blocks 501 to 507.
  • the tape extender 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 described above 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 broken down 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 transformation without windowing (or equivalently with an implicit rectangular window of the length of the frame) is possible when the processing is carried out in the domain of the excitation, and not the domain of the signal. 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 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, pp. 144-149 , and implemented in ITU-T G.718 Annex B and G.729.1 Annex E.
  • EDCT Evolved DCT
  • the DCT-IV transformation can be replaced by other short-term time-frequency transformations of the same length and in the field of excitation or in the field of the signal, like 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 Tranform” in English
  • the delay T in the block 310 of the figure 3 should be adjusted (reduced) adequately as a function of the additional delay due to the analysis / synthesis by this transform.
  • the decomposition into sub-bands is carried out by applying 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 performing 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 surround 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 figure 4 , that is to say the extension of the decoded low band signal.
  • the original spectrum is preserved, in order to be able to apply a progressive attenuation response of the high-pass filter to it in this frequency band and also 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) greater than the first frequency band. (0-6.4 kHz).
  • the extension of the decoded low band signal takes place at least over the second frequency band but also over 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 set 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 go beyond the scope of the invention without changing its scope.
  • the high band (> 6 kHz) contains surround information which is naturally similar to that present in the low band.
  • the ambience is defined 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 generally correlates with that of the 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 a surround signal implements the step E402 of the figure 4 in the frequency domain.
  • 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. The extraction of this ambient signal could also be done on the decoded excitation but not extended, that is to say before the step of spectral extension or translation, that is to say for example on a portion of the low frequency signal rather than directly on the high frequency signal.
  • 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 surround signal and the tonal components.
  • an energy level control factor is calculated based on the total energy of the decoded (or decoded and extended) low band signal and tonal components.
  • ⁇ makes it possible to avoid an overestimation of the energy.
  • ⁇ is calculated so as to keep the same level of the ambient signal with respect to the energy of the tonal components in the consecutive bands of the signal.
  • 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 ⁇ may be replaced by other methods.
  • we can extract (calculate) various parameters (or “features” in English) characterizing the signal in low band, including a “tilt” parameter similar to that calculated in the AMR-WB codec, and we will estimate the postman ⁇ as a function of a linear regression from these different parameters by limiting its value between 0 and 1.
  • the linear regression can for example be estimated in a supervised way by estimating the factor ⁇ by giving the original high band in a base d 'learning. It will 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 in a particular embodiment optionally performs a double operation of applying the band-pass filter frequency response and de-emphasis (or de-emphasis) filtering ) in the frequency domain.
  • the de-emphasis filtering can be carried out in the time domain, after block 502 or even before block 510; however, in this case, the bandpass filtering performed in block 501 may leave some low frequency components of very low levels which are amplified by de-emphasis, which may slightly noticeably alter the decoded low band. 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 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 the HF synthesis.
  • the de-emphasis may be performed in an equivalent manner in the time domain after reverse DCT.
  • bandpass filtering is applied with two separate parts: one fixed high pass, the other adaptive low pass (rate dependent).
  • This filtering is performed in the frequency domain.
  • the band-pass filtering can be adapted by defining a single filtering step combining the high-pass and low-pass filtering.
  • the band-pass filtering could be performed in an equivalent manner in the time domain (as in block 112 of the figure 1 ) with different filter coefficients depending on the flow, after a reverse DCT step.
  • it is advantageous to carry out this step directly in the frequency domain because 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. .
  • the block 502 carries out the synthesis corresponding to the analysis carried out in the block 510.
  • the signal sampled at 16 kHz is then optionally scaled by gains defined by subframe of 80 samples (block 504).
  • block 503 differs from that of block 101 of figure 1 , because the energy at the level of the current frame is taken into account in addition to that of the sub-frame. This makes it possible to have the ratio of the energy of each sub-frame 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 keep the energy ratio between the sub-frame and the frame in the high band 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 changes to 20 at the rate of 6.6, which does not significantly change the quality of the synthesized signal.
  • the LPC synthesis filtering could be carried out in the frequency domain, after having calculated the frequency response of the filter implemented in the block 507.
  • the encoding of the low band (0-6.4 kHz) could be replaced by a CELP encoder other than that used in AMR-WB, such as for example the CELP encoder in G.718 to 8 kbit / s.
  • a CELP encoder other than that used in AMR-WB, such as for example the CELP encoder in G.718 to 8 kbit / s.
  • other wideband encoders or encoders operating at frequencies above 16 kHz in which the low band encoding 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 low-band decoding does not use linear prediction, there is no excitation signal to extend, in this case an LPC analysis of the signal reconstructed in the current frame can be performed and an LPC excitation will be calculated. so as to be able to apply the invention
  • the excitation or the low band signal ( u ( n )) is resampled, 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 drawback of being more complex, because the transform (DCT-IV) of the excitation or of the signal is then calculated on a larger length and resampling is not performed in the transform domain.
  • the figure 6 shows an exemplary hardware embodiment of a tape extension device 600 according to the invention. This can be an integral part of an audio-frequency signal decoder or of equipment receiving audio-frequency signals, whether or not decoded.
  • This type of device comprises a processor PROC cooperating with a memory block BM comprising a storage and / or working memory MEM.
  • a device comprises an input module E capable of receiving a decoded or extracted audio signal in a first frequency band called the low band brought back to 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 filtering module 501 of the figure 5 .
  • the memory block can advantageously comprise a computer program comprising code instructions for the implementation of 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 extracting (E402) tonal components and a surround signal from a signal from the decoded low-band signal ( U ( k )) , combining (E403) tonal components (y (k)) and the surround signal (U HBA ( k )) by adaptive mixing using energy level control factors to obtain an audio signal, called the combined signal ( U HB 2 ( k )), of 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 combining step.
  • 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 in 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 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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Claims (9)

  1. Verfahren zum Erweitern des Frequenzbandes eines Audiofrequenzsignals während eines Decodierungs- oder Verbesserungsprozesses, umfassend einen Schritt zum Erhalten des decodierten Signals in einem ersten Frequenzband, das als Niederband bekannt ist, wobei das Verfahren umfasst die folgenden Schritte, die im Frequenzbereich ausgeführt werden:
    - Extraktion (E402) von Tonkomponenten und eines Umgebungssignals aus einem Signal aus dem decodierten Niederbandsignal;
    - Kombination (E403) der Tonkomponenten und des Umgebungssignals durch adaptives Mischen unter Verwendung von Energiepegelsteuerungsfaktoren, um ein kombiniertes Signal zu erhalten;
    - Erweiterung (E401a) über mindestens ein zweites Frequenzband, das höher ist als das erste Frequenzband des decodierten Niederbandsignals vor dem Extraktionsschritt oder des kombinierten Signals nach dem Kombinationsschritt;
    - Synthese (E404b) eines Audiosignals, um es in den Zeitbereich zu bringen:
    - das kombinierte Signal, wenn der Erweiterungsschritt (E401a) vor dem Extraktionsschritt (E402) ausgeführt wird, oder
    - das erweiterte kombinierte Signal, wenn der Erweiterungsschritt (E401a) nach dem Kombinationsschritt (E403) ausgeführt wird;
    wobei das Verfahren dadurch gekennzeichnet ist, dass der Schritt des Extrahierens der Tonkomponenten und des Umgebungssignals gemäß den folgenden Schritten ausgeführt wird:
    - Erhalten des Umgebungssignals durch Berechnen eines Durchschnittswerts des Spektrums des decodierten oder decodierten und erweiterten Niederbandsignals;
    - Erhalten der Tonkomponenten durch Subtrahieren des berechneten Umgebungssignals von dem decodierten oder decodierten und erweiterten Niederbandsignal.
  2. Verfahren nach Anspruch 1, dadurch gekennzeichnet, dass das decodierte Niederbandsignal ein decodiertes Niederbandanregungssignal ist.
  3. Verfahren nach Anspruch 1, dadurch gekennzeichnet, dass ein für das adaptive Mischen verwendeter Energieniveauregelfaktor, basierend auf der Gesamtenergie der decodierten oder decodierten und erweiterten Niederbandsignal- und Tonkomponenten, berechnet wird.
  4. Verfahren nach einem der vorangehenden Ansprüche, dadurch gekennzeichnet, dass das decodierte Niederbandsignal einem Schritt der Zerlegung durch Transformation oder Filterbank in Teilbänder unterliegt, wobei die Extraktions- und Kombinationsschritte dann im Frequenzbereich oder in Teilbändern ausgeführt werden.
  5. Verfahren nach einem der vorangehenden Ansprüche, dadurch gekennzeichnet, dass der Erweiterungsschritt des decodierten Niederbandsignals nach der folgenden Gleichung ausgeführt wird: U HB 1 k = { 0 k = 0 , ⋯ , 199 U k k = 200 , ⋯ , 239 U k + start _ band − 240 k = 240 , ⋯ , 319
    Figure imgb0046
     wobei k der Index der Probe, U(k) das Spektrum des decodierten Niederbandsignals ist, das nach einem transformierten Schritt erhalten wurde, UHB1(k) das Spektrum des erweiterten Signals und start_band eine vordefinierte Variable ist.
  6. Vorrichtung zum Erweitern des Frequenzbandes eines Audiofrequenzsignals, wobei das Signal in einem ersten Frequenzband decodiert wurde, das als Niederband bezeichnet wird, wobei die Vorrichtung umfasst:
    - ein Modul (512) zum Extrahieren von Tonkomponenten und eines Umgebungssignals im Frequenzbereich aus einem Signal, das von dem decodierten Niederbandsignal stammt;
    - ein Modul (513) zum Kombinieren von Tonkomponenten und des Umgebungssignals im Frequenzbereich durch adaptives Mischen unter Verwendung von Energiepegelsteuerfaktoren, um ein kombiniertes Signal zu erhalten;
    - ein Erweiterungsmodul (511) im Frequenzbereich in mindestens einem zweiten Frequenzband, das größer ist als das erste Frequenzband, das auf dem decodierten Niederbandsignal vor dem Extraktionsmodul oder auf dem kombinierten Signal nach dem Kombinationsmodul implementiert ist;
    - ein Synthesemodul (502) eines Audiosignals, um es in den Zeitbereich zurückzubringen:
    • das kombinierte Signal, wenn das Erweiterungsmodul (511) vor dem Extraktionsmodul (512) aktiviert ist, oder
    • das kombinierte Signal verlängert wird, wenn das Erweiterungsmodul (511) nach dem Kombinationsmodul (513) aktiviert wird;
    wobei die Vorrichtung dadurch gekennzeichnet ist, dass das Modul zum Extrahieren der Tonkomponenten und des Umgebungssignals konfiguriert ist
    - das Umgebungssignals durch Berechnen eines Durchschnittswerts des Spektrums des decodierten oder decodierten und erweiterten Niederbandsignals zu erhalten;
    - Tonkomponenten durch Subtraktion eines berechneten Umgebungssignals vom decodierten oder decodierten und erweiterten Niederbandsignal zu erhalten.
  7. Audiofrequenz-Signaldecoder, dadurch gekennzeichnet, dass er eine Frequenzband-Erweiterungsvorrichtung nach Anspruch 6 umfasst.
  8. Computerprogramm mit Codebefehlen zum Implementieren der Schritte des Frequenzband-Erweiterungsverfahrens nach einem der Ansprüche 1 bis 5, wenn diese Anweisungen von einem Prozessor ausgeführt werden.
  9. Speichermedium, das von einer Frequenzband-Erweiterungsvorrichtung lesbar ist, auf der ein Computerprogramm aufgezeichnet ist, das Code-Anweisungen zur Ausführung der Schritte des Frequenzband-Erweiterungsverfahrens gemäß einem der Ansprüche 1 bis 5 umfasst.
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PL15705687T PL3103116T3 (pl) 2014-02-07 2015-02-04 Ulepszone rozszerzanie pasma częstotliwości w dekoderze sygnałów o częstotliwości akustycznej
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RS20210945A RS62160B1 (sr) 2014-02-07 2015-02-04 Poboljšana ekstenzija frekvencijskog opsega u dekoderu audio frekventnih signala
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MX2016010214A (es) 2016-11-15
RU2763481C2 (ru) 2021-12-29
PL3327722T3 (pl) 2024-07-15
KR20160119150A (ko) 2016-10-12
ZA201708368B (en) 2018-11-28
JP2019168709A (ja) 2019-10-03
US10043525B2 (en) 2018-08-07
US10668760B2 (en) 2020-06-02
RU2763547C2 (ru) 2021-12-30
ES2955964T3 (es) 2023-12-11
WO2015118260A1 (fr) 2015-08-13
RU2017144521A3 (de) 2021-04-01
RU2016136008A (ru) 2018-03-13
RU2016136008A3 (de) 2018-09-13
CN105960675A (zh) 2016-09-21
CN108022599A (zh) 2018-05-11
RU2017144521A (ru) 2019-02-18
US20180304659A1 (en) 2018-10-25
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JP6775064B2 (ja) 2020-10-28
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CN108109632B (zh) 2022-03-29
EP3330967B1 (de) 2024-04-10
RU2763848C2 (ru) 2022-01-11
US11312164B2 (en) 2022-04-26
EP3330967A1 (de) 2018-06-06
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EP3330966B1 (de) 2023-07-26
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US20200353765A1 (en) 2020-11-12
DK3103116T3 (da) 2021-07-26
JP6775065B2 (ja) 2020-10-28
LT3330966T (lt) 2023-09-25
HUE055111T2 (hu) 2021-10-28
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