EP3014611A1 - 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 audiofréquencesInfo
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- EP3014611A1 EP3014611A1 EP14742262.0A EP14742262A EP3014611A1 EP 3014611 A1 EP3014611 A1 EP 3014611A1 EP 14742262 A EP14742262 A EP 14742262A EP 3014611 A1 EP3014611 A1 EP 3014611A1
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- extended
- frequency
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Classifications
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- 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
- G10L21/0388—Details of processing therefor
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- 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
- G10L19/012—Comfort noise or silence coding
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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
- G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
- G10L19/04—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using predictive techniques
- G10L19/06—Determination or coding of the spectral characteristics, e.g. of the short-term prediction coefficients
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- 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
- G10L19/08—Determination or coding of the excitation function; Determination or coding of the long-term prediction parameters
- G10L19/083—Determination or coding of the excitation function; Determination or coding of the long-term prediction parameters the excitation function being an excitation gain
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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
- 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
- G10L19/08—Determination or coding of the excitation function; Determination or coding of the long-term prediction parameters
- G10L19/12—Determination or coding of the excitation function; Determination or coding of the long-term prediction parameters the excitation function being a code excitation, e.g. in code excited linear prediction [CELP] vocoders
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- 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
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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
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- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
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- 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
- G10L19/08—Determination or coding of the excitation function; Determination or coding of the long-term prediction parameters
Definitions
- the present invention relates to the field of coding / decoding and audio-frequency signal processing (such as speech, music or other signals) for their transmission or storage.
- the invention relates to a method and a device for extending the frequency band in a decoder or a processor performing an audio-frequency signal improvement.
- state of the art audio signal coding (mono) consists of perceptual encoding by transform or subband, with parametric coding of high frequencies by tape replication.
- 3GPP AMR-WB Adaptive Multi-Rate Wideband
- codec and decoder 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 parametrically reconstructed by "band extension" ( or BWE for "Bandwidth Extension” with or without additional information depending on the mode of the current frame.
- BWE Bandwidth Extension
- the limitation of the coded band of the AMR-WB codec at 7 kHz is essentially related to the fact that the transmission frequency response of the broadband terminals has been approximated at the time of standardization (ETSI / 3GPP then ITU-T T) according to the frequency mask defined in the ITU-T P.341 standard and more precisely by using a so-called "P341" filter defined in the ITU-T G.191 standard which cuts frequencies above 7 kHz (this filter respects the mask defined in P.341).
- a signal sampled at 16 kHz may have a defined audio band of 0 to 8000 Hz; the AMR-WB codec thus introduces a limitation of the high band in comparison with the theoretical bandwidth of 8 kHz.
- the 3GPP AMR-WB speech codec was standardized in 2001 mainly for circuit-mode (CS) telephony applications on GSM (2G) and UMTS (3G). This same codec was also standardized in 2003 in ITU-T as Recommendation G.722.2 "Wideband coding speech at around 16kbit / s using Adaptive Multi-Rate Wideband (AMR-WB)".
- AMR-WB coding and decoding algorithm The details of the AMR-WB coding and decoding algorithm are not repeated here, a detailed description of this codec is found in the 3GPP specifications (TS 26.190, 26.191, 26.192, 26.193, 26.194, 26.204) and ITU-TG .722.2 (and the corresponding Appendices and Appendix) and in the article by B. Bessette et al. entitled "The adaptive multirate broadband speech coded (AMR-WB)", IEEE Transactions on Speech and Audio Processing, vol. 10, no. 8, 2002, pp. 620-636 and associated 3GPP and ITU-T standard source codes.
- AMR-WB adaptive multirate broadband speech coded
- the principle of band extension in the AMR-WB codec is rather rudimentary. Indeed, the high band (6.4-7 kHz) is generated by formatting 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 band extension technique is illustrated in Figure 1.
- a white noise, u HB1 (n), n 0, ⁇ ⁇ ⁇ , 79, is generated at 16 kHz per 5 ms subframe per linear congruent generator (block 100).
- This noise u HB1 (n) is shaped in time by applying gains per subframe; this operation is broken down into two processing steps (blocks 102, 106 or 109):
- u HB (n) g HB u HB2 (n)
- SHB W SPSSP + ( 1 - W SP) SBG
- w sp is a weighting function that depends on Voice Activity Detection (VAD).
- VAD Voice Activity Detection
- correction information is transmitted by the encoder AMR-WB and decoded (blocks 107, 108) in order to refine the estimated gain per subframe (4 bits every 5ms, ie 0.8 kbit / s) .
- the artificial excitation u HB (n) is then filtered (block 111) by a transfer function function LPC (block 111) synthesis filter 1 / A HB (z) and operating at the sampling frequency of 16 kHz.
- LPC transfer function function
- the filter 1 / A HB (z) is of order 16 and simply corresponds to:
- a low-pass filter also FIR type (block 113) is added to the treatment to further attenuate frequencies above 7 kHz.
- the synthesis at high frequencies (HF) is finally added (block 130) to the low frequency synthesis (BF) obtained with the blocks 120 to 123 and resampled at 16 kHz (block 123).
- HF high frequencies
- BF low frequency synthesis
- the signal in the high band is white noise formatted (by temporal gains per subframe, filtering by 1 / A HB (z) and bandpass filtering), which is not a good general model signal in the band 6.4-7 kHz.
- white noise formatted by temporal gains per subframe, filtering by 1 / A HB (z) and bandpass filtering
- bandpass filtering which is not a good general model signal in the band 6.4-7 kHz.
- there are very harmonic music signals for which the 6.4-7 kHz band contains sinusoidal components (or tones) and no noise (or little noise) for these signals the band extension of the AMR-WB coding degrades. strongly the quality.
- the 7 kHz low-pass filter (block 113) introduces an offset of nearly 1 ms between the low and high bands, which can potentially degrade the quality of some signals by slightly desynchronizing the two bands at 23.85 kbit / s - this Desynchronization can also be a problem when switching from 23.85 kbit / s to other modes.
- the estimate of gains per subframe (block 101, 103 to 105) is not optimal. In part, it is based on an equalization of the "absolute" energy per sub-frame (block 101) between signals at different frequencies: the artificial excitation at 16 kHz
- the AMR-WB decoding algorithm has been improved in part with the development of the ITU-T G.718 scalable codec which was standardized in 2008.
- ITU-T G.718 includes an interoperable mode, for which core coding is compatible with 12.65 kbit / s G.722.2 (AMR-WB) coding; in addition, the G.718 decoder has the particularity of being able to decode a bit stream AMR-WB / G.722.2 at all possible bit rates of the AMR-WB codec (from 6.6 to 23.85 kbit / s).
- the G.718 interoperable decoder in low delay mode (G.718-LD) is illustrated in FIG. 2.
- G.718-LD low delay mode
- the band extension (described 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 band-pass filter and the filter Synthesis 1 / A HB (z) (Blocks 111 and 112) are in reverse order.
- the 4 bits transmitted by AMR-WB encoder subframes 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.
- the low-pass filter at 7 kHz (block 113) is not used, and the specific decoding mode 23.85 kbit / s is omitted (blocks 107 to 109).
- a post-processing of the 16 kHz synthesis is implemented in G.718 by "noise facede” in block 208 (to “improve” the quality of the silences by reducing the level ), high-pass filtering (block 209), low-frequency post-filter (so-called “bass posfilter”) in block 210 attenuating interharmonic noise at low frequencies, and conversion to 16-bit integers with saturation control (with control of gain or AGC) in block 211.
- the present invention improves the situation.
- the invention proposes a method of extending the frequency band of an audiofrequency signal during a decoding or improvement process comprising a decoding or extraction step, in a so-called first frequency band. low band, an excitation signal and the coefficients of a linear prediction filter.
- the method is such that it comprises the following steps:
- the excitation signal (resulting from the decoding of the low band or from a low band signal extraction) makes it possible to carry out the band extension with a signal model that is more suitable for certain types of band. signals like music signals.
- the excitation signal decoded or estimated in the low band has in some cases harmonics, which when they exist, can be transposed in high frequency so that it ensures a certain level of harmonicity in the high band reconstructed.
- the band extension according to the method thus makes it possible to improve the quality for this type of signal.
- the band extension according to the method is performed by first extending an excitation signal and then applying a synthesis filtering step; this approach exploits the fact that the decoded excitation in the low band is a signal whose spectrum is relatively flat, which avoids the decoded signal whitening treatments that may exist in the known methods of band extension in the frequency domain in the state of the art.
- first frequency band the energy at the current frame and that of the subframe in the low band signal
- second frequency band the energy per frame in the high band
- This makes it possible to keep in the high band the same ratio of energy between subframe and frame as in the low band, which is particularly beneficial when the energy of the subframes varies greatly, for example in the case of transient sounds. , of attacks.
- the method further includes an adaptive bandpass filtering step based on the decoding rate of the current frame.
- This adaptive filtering makes it possible to optimize the extended bandwidth as a function of the bit rate, and therefore the quality of the reconstructed signal after band extension. Indeed, for low bit rates (typically 6.6 and 8.85 kbit / s for AMR-WB), the overall quality of the decoded signal in low band (by AMR-WB codec or an interoperable version) not being very good, it is better not to extend the decoded band too much and thus to limit the band extension by adapting the frequency response of the filter bandpass associated to cover for example an approximate band of 6 to 7 kHz; this limitation is all the more advantageous that the excitation signal itself is relatively poorly coded and it is preferable not to use a sub-band too wide for the extension of high frequencies.
- the quality can be improved with HF synthesis covering a wider band, for example approximately 6 to 7.7 kHz.
- the high limit of 7.7 kHz (instead of 8 kHz) is an example embodiment, which can be adjusted to values close to 7.7 kHz. This limitation is here justified by the fact that the extension is made in the invention without auxiliary information and that an extension up to 8 kHz (although theoretically possible) could result in artifacts for particular signals.
- this limitation at 7.7 kHz takes into account the fact that typically the anti-aliasing filters in analog / digital conversion and resampling filters between 16 kHz and other frequencies are not perfect and they typically introduce a rejection at frequencies below 8 kHz.
- the method comprises a step of transforming the time-frequency of the excitation signal, the step of obtaining an extended signal then taking place in the frequency domain and a reverse time-frequency transforming step. of the extended signal before the scaling and filtering steps.
- the implementation of the band extension (of the excitation signal) 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 a resolution Frequency sufficient to detect harmonics and transpose high harmonics of the signal (in the low band) to improve quality while respecting the structure of the signal.
- the step of generating an oversampled and extended excitation signal is performed according to the following equation:
- this function includes a resampling of the excitation signal by adding samples to the spectrum of this signal.
- the original spectrum is conserved, in order to be able to apply a progressive attenuation response of the high-pass filter in this frequency band and also to not introduce defects. audible during the step of adding the low frequency synthesis to the high frequency synthesis.
- the method comprises a de-emphasis filtering step of the extended signal at least in the second frequency band.
- the signal in the second frequency band is brought into a domain coherent with the signal in the first frequency band.
- the method further comprises a step of generating a noise signal at least in the second frequency band, the extended signal being obtained by combining the extended excitation signal and the noise signal.
- the combining step is performed by adaptive additive mixing with a level equalization gain between the extended excitation signal and the noise signal.
- this equalization gain allows the combining step to adapt to the characteristics of the signal to optimize the relative proportion of noise in the mixture.
- the present invention also aims at a frequency band extension device for an audio frequency signal comprising a decoding or extraction stage, in a first so-called low band frequency band, of an excitation signal and of the transmission coefficients.
- a linear prediction filter is such that it comprises:
- a filtering module (510) of said extended signal scaled by a linear prediction filter whose coefficients are derived from the coefficients of the low band filter.
- 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. It is directed to a computer program comprising code instructions for implementing the steps of the tape 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 to the band expansion device, possibly removable, storing a computer program implementing a band extension method as described above.
- FIG. 1 illustrates a part of an AMR-WB decoder implementing frequency band extension steps of the state of the art and as previously described;
- FIG. 2 illustrates a decoder of the interoperable type G.718-LD at 16 kHz according to the state of the art and as described previously;
- FIG. 3 illustrates an interoperable decoder with the AMR-WB coding and integrating a band extension device according to one embodiment of the invention
- FIG. 4 illustrates in flowchart form the main steps of a band extension method according to one embodiment of the invention
- FIG. 5 illustrates a first embodiment in the frequency domain of a band extension device according to the invention
- FIG. 6 illustrates an example of frequency response of a bandpass filter used in a particular embodiment of the invention
- FIG. 7 illustrates a second embodiment in the time domain of a band extension device according to the invention.
- FIG. 8 illustrates a hardware embodiment of a band extension device according to the invention.
- FIG. 3 illustrates an exemplary decoder, compatible with the AMR-WB / G.722.2 standard, in which there is a postprocessing similar to that introduced in G.718 and described with reference to FIG. 2 and an improved band extension according to the extension method of the invention, implemented by the band extension device illustrated by block 309.
- the 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) being the subject of the invention operates at the frequency of 16 kHz
- the synthesis BF and HF are combined (block 312) at the frequency fs after adequate resampling (block 306 and internal processing block 311).
- the combination of the low and high bands can be done at 16 kHz, after resampling the low band of 12.8 to 16 kHz, before resampling the extended signal at the frequency fs.
- the decoding according to FIG. 3 depends on the mode (or bit rate) AMR-WB associated with the current frame received.
- the decoding of the low band CELP part comprises the following steps:
- the post-treatments applied to the excitation can be modified (for example, the phase dispersion can be improved) or these post
- the treatments may be extended (for example, interharmonic noise reduction may be implemented) without affecting the nature of the band extension method of the invention.
- the decoding of the low band described above assumes a current frame called "active" with a rate between 6.6 and 23.85 kbit / s.
- active a current frame
- some 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 describes several parameters: ISF parameters averaged over 8 frames, average energy over 8 frames, dithering flag for the non-stationary noise reconstruction.
- the decoder makes it possible to extend the decoded low band (50-6400 Hz while taking into account the 50 Hz high-pass filtering at the decoder, 0-6400 Hz in the decoder. 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 extension of the excitation is carried out in the frequency domain in a band of 5000 to 8000 Hz, to allow bandpass filtering of width 6000 to 6900 or 7700 Hz.
- the gain correction information HF (0.8 kbit / s) transmitted at 23.85 kbit / s is here ignored.
- FIG. 3 no specific block at 23.85 kbit / s is used.
- the high band decoding part is produced in the block 309 representing the band extension device according to the invention and which is detailed in FIG. 5 in a first embodiment and in FIG. 7 in a second embodiment.
- This device comprises at least one module for obtaining an extended signal in at least one second frequency band greater than the first frequency band from an oversampled excitation signal and extended in at least a second band.
- of frequency U H Bi (k)
- a delay (block 310) is introduced in the first embodiment to synchronize the outputs of the blocks 306 and 307 and the high band synthesized at 16 kHz is resampled from 16 kHz to the frequency fs (block output 311).
- the delay ⁇ 30 samples, which corresponds to the resampling delay of 12.8 to 16 kHz of 15 samples + delay of the low frequency post-processing of 15 samples.
- the extension method of the invention implemented in block 309 according to the first embodiment introduces preferentially no 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 value of in block 310 will have to be adjusted according to the specific implementation.
- T is reduced to compensate for the delay introduced by the post-processing of the low frequencies (block 306) if it is used.
- the low and high bands are then combined (added) in block 312 and the resulting synthesis is post-processed by high-order 50 Hz (type IIR) high-pass filtering whose coefficients depend on the frequency fs (block 313) and output post-processing with optional noise gate application 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 FIG. 3, implements a band extension method described now with reference to FIG. 4.
- This extension device may also be independent of the decoder and may implement the method described in FIG. 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 it an excitation and an LPC filter.
- This device receives as input an excitation signal in a first so-called low-band frequency band u (n) in the case of an implementation in the time domain or U (k) in the case of an implementation. in the frequency domain for which a time-frequency transform step is then applied.
- this received excitation signal is a decoded signal.
- the low band excitation signal is extracted by analysis of the audio signal.
- the low band audio signal is resampled before the excitation extraction step, so that the excitation extracted from the audio signal by linear prediction estimated from the low band signal (or parameters LPC associated with the low band) is already resampled.
- An exemplary embodiment in this case consists in taking a sampled low band signal at 12.8 kHz, which has a low-band LPC filter describing the short-term spectral envelope for the current frame, oversampling it at 16 kHz, and filter it by an LPC prediction filter obtained by extrapolating the LPC filter.
- Another embodiment is to take a low band signal sampled at 12.8 kHz which is not available LPC model, the oversampler at 16 kHz, perform an LPC analysis on this signal at 16 kHz, and filter this signal by a LPC prediction filter obtained by this analysis.
- a step E401 for generating an extended oversampled excitation signal (e xt (n) or U HB1 (k)) in a second frequency band greater than the first frequency band is performed.
- This generation step may comprise both a resampling step and an extension step or simply an extension step depending on the excitation signal obtained at the input.
- This step is detailed later in the embodiments described with reference to FIGS. 5 and 7.
- This extended oversampled excitation signal is used to obtain an extended signal (U H B2 (k)) in a second frequency band.
- This extended signal then has a signal model adapted to certain types of signals thanks to the characteristics of the extended excitation signal.
- This extended signal can be obtained after combining the oversampled and extended excitation signal with another signal, for example a noise signal.
- a step E402 for generating a noise signal (u HB (n) or U HB (k)) at least in the second frequency band is performed.
- the second frequency band is for example a high frequency band ranging from 6000 to 8000 Hz.
- this noise can be generated pseudo-randomly by a linear congruential generator.
- this noise generation can be replaced by other methods, for example a signal of constant amplitude (of arbitrary value such as 1) could be defined and random signs applied to each frequency line. generated.
- the extended excitation signal is then combined with the noise signal in step E403 to obtain the extended signal which may also be called a combined signal (u HB1 (n) or U HB2 (k)) in the frequency band range corresponding to the entire frequency band including the first and the second frequency band.
- a combined signal u HB1 (n) or U HB2 (k)
- the combination of these two types of signals makes it possible to obtain a combined signal with characteristics more adapted to certain types of signals such as musical signals.
- the decoded or estimated excitation signal in the low band in some cases has harmonics closer to the musical signals than the noise signal alone.
- Low-frequency harmonics if they exist, can thus be transposed into high frequency so that their mixing with noise makes it possible to ensure a certain level of harmonicity or relative level of noise or spectral flatness ("spectral flatness"). in English) in the reconstructed high band.
- the band extension according to the method improves the quality for this type of signals compared to AMR-WB.
- the combined (or extended) signal is then filtered at E404 by a linear prediction filter whose coefficients are derived from the coefficients of the low band filter (((z)) decoded or obtained by analysis and extraction from the low band signal or an oversampled version of it.
- the band extension according to the method is therefore achieved by first extending an excitation signal and then applying a linear prediction synthesis filtering step (LPC); this approach exploits the fact that the LPC decoded excitation in the low band is a signal whose spectrum is relatively flat, which avoids additional processing of whitening of the decoded signal in the band extension.
- the coefficients of this filter can for example be obtained from the decoded parameters of the linear band prediction filter (LPC).
- the LPC filter used in the high sampled 16 kHz band is of the form ⁇ l A ⁇ zl ⁇ ), where 1 / ⁇ (z) is the low band decoded filter, and ⁇ is a weighting factor, the frequency response of the IIA filter (zl ⁇ ) corresponds to a spread of the frequency response of the decoded low band filter.
- the filter 1 / A (z) can be extended to a higher order (as to 6.6 kbit / s in the block 111) to avoid such spreading.
- additional steps of adaptive bandpass filtering at E405 and / or scaling at E406 and E407 can be performed to improve the quality of the extension signal according to the bit rate. decoding and on the other hand to ensure to keep the same energy ratio between a subframe and a combined signal frame as in the low frequency band.
- the band extension device is described now with reference to FIG. 5. This device implements the band extension method described above with reference to FIG. 4.
- a low band excitation signal decoded or estimated by analysis is received (u (n)).
- the band extension uses the decoded excitation at 12.8 kHz (exc2 or u (n)) at the output of block 302.
- the generation of the oversampled and extended excitation is carried out in a frequency band ranging from 5 to 8 kHz including a second frequency band (6.4-8kHz) greater than the first band of frequency (0-6.4 kHz).
- the generation of the extended excitation signal is effected at least on the second frequency band but also on a part of the first frequency band.
- this signal is transformed to obtain an excitation signal spectrum U (k) by the time-frequency transformation module 500.
- the DCT-IV transformation is implemented by FFT according to the "Evolved ZXT (EDCT)" algorithm described in the article by D. M. 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 ZXT
- the DCT-IV transformation may be replaced by other short-term time-frequency transformations of the same length and in the field of excitation, such as an FFT (for Fast Fourier Transform "in English) or DCT-II (Discrete Cosine Transform - Type II).
- FFT Fast Fourier Transform
- DCT-II Discrete Cosine Transform - Type II
- MDCT for "Modified Discrete Cosine Tranform"
- This approach preserves the original spectrum in this band and avoids introducing distortions in the 5000-6000 Hz band during the addition of HF synthesis with BF synthesis - particularly the signal phase (implicitly represented in the DCT-IV domain) in this band is preserved.
- the 6000-8000 Hz band of U HB1 (k) is here defined 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 changing the nature of the invention.
- the details of the adaptation of the value start_band are not described here because they go beyond the scope of the invention without changing the scope.
- the noise in the 6000-8000 Hz band
- the noise is generated pseudo-randomly with a 16-bit linear congruent generator
- the combination block 503 can be realized in different ways. In a privileged way, we consider an adaptive additive mix of the form:
- G HBN is a normalization factor for equalizing the energy level between the two signals
- the coefficient (between 0 and 1) is adjusted according to parameters estimated from the decoded low band and the coefficient ⁇ (between 0 and 1) depends on a.
- the energy of the noise is calculated in three bands: 2000-4000 Hz, 4000-6000 Hz and 6000-8000 Hz, with
- N ( ⁇ , ⁇ 2 ) is the set of indices k for which the index coefficient est is classified as being associated with noise. This set can be obtained for example by detecting the local peaks in U and in whereas these lines are not associated with noise, ie (by applying the negation of the previous condition):
- N (a, b) ⁇ a ⁇ k ⁇ b ⁇ U + 1)
- the calculation of a may be replaced by other methods.
- the linear regression could for example be estimated in a supervised manner by estimating the factor a by giving itself the original high band in a learning base. Note that the calculation method does not limit the nature of the invention.
- the factors ⁇ and a may be adapted to take account of the fact that noise injected into a given band of the signal is generally perceived as stronger than a harmonic signal at the same energy in the same band.
- the block 503 realizes the equivalent of the block 101 of FIG. 1 to normalize the white noise as a function of an excitation which is on the other hand here in the frequency domain, already extended at the rate of 16 kHz; in addition, the mix is limited to the band 6000-8000 Hz.
- the block 504 optionally carries out a dual operation of application of bandpass filter frequency response and deemphasis filtering (or deemphasis) in the frequency domain.
- the deemphasis filtering may be performed in the time domain, after block 505 or even before block 500; however, in this case, bandpass filtering performed in block 504 may leave some low frequency components of very low levels which are amplified by de-emphasis, which may slightly discern the decoded low band. For this reason, it is preferred here to perform the deemphasis in the frequency domain.
- G deem ⁇ (k) is the frequency response of the filter l / (l - 0.68z _1 ) over a restricted discrete frequency band. Taking into account the discrete (odd) frequencies of the DCT-IV, we define here G deem ⁇ (k) as:
- the definition of 6 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 to a domain coherent with the low frequency signal (0-6.4 kHz) coming out of block 305. This is important for the estimation and the subsequent adjustment of the energy of the HF synthesis.
- the de-emphasis can be performed in an equivalent way in the time domain after inverse DCT.
- Such an embodiment is implemented in Figure 7 described below.
- band-pass filtering is applied with two separate parts: one fixed high-pass, the other adaptive low-pass (flow-rate function).
- the 3 dB cutoff frequencies are 6000 Hz for the low part and for the high part approximately 6900, 7300, 7600 Hz at 6.6, 8.86 and at rates greater than 8.85 kbit / s (respectively).
- the partial low-pass filter response in the frequency domain is calculated as follows:
- G hp (k) may be modified while keeping a gradual attenuation.
- the variable bandwidth low-pass filtering, G lp (k) may be adjusted with different values or frequency support, without changing the principle of this filtering step.
- the bandpass filtering example illustrated in FIG. 6 can be adapted by defining a single filtering step combining the high-pass and low-pass filterings.
- the bandpass filtering may be performed equivalently in the time domain (as in block 112 of FIG. 1) with different filter coefficients according to the bit rate, after an inverse DCT step.
- Such an embodiment is implemented in Figure 7 described below.
- it is advantageous to carry out this step directly in the frequency domain because the filtering is carried out in the field of LPC excitation and therefore the problems of circular convolution and edge effects are very limited in this field. .
- the inverse transform block 505 performs an inverse DCT on 320 samples to find the high frequency excitation sampled at 16 kHz. Its implementation is identical to block 500, because the DCT-IV is orthonormed, except that the length of the transform is 320 instead of 256, and we obtain:
- This excitation sampled at 16 kHz is then optionally scaled by gains defined by subframe of 80 samples (block 507).
- a gain g H Bi (m) per subframe is first calculated (block 506) by energy ratios of the subframes such that in each subframe the frame common:
- Block 507 scales the combined (or extended) signal (step E406 of FIG. 4) according to the following equation:
- the embodiment of the block 506 differs from that of the block 101 of Figure 1, because the energy at 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 sub-frame with respect to the energy of the frame. Energy ratios (or relative energies) are compared rather than the absolute energies between low band and high band.
- this scaling step makes it possible to keep in the high band the energy ratio between the subframe and the frame in the same way as in the low band.
- block 509 then scales the signal (step E407 of FIG. 4) according to the following equation:
- the gain g HB2 (m) is obtained from the block 508 by executing the blocks 103, 104 and 105 of the AMR-WB coding (the input of the block 103 being the decoded excitation in the low band, u (n)) .
- Blocks 508 and 509 are useful for adjusting the level of the LPC synthesis filter (block 510), here depending on the tilt of the signal. Other methods of calculating the gain g HB2 (m) are possible without changing the nature of the invention.
- this filtering can be done in the same way as described for the block 111 of FIG. 1 of the AMR-WB decoder, however the order of the filter goes to 20 at the rate of 6.6, which does not change. not significantly the quality of the synthesized signal.
- the coding of the low band (0-6.4 kHz) may be replaced by a CELP coder other than that used in AMR-WB, for example the CELP coder in G.718 to 8. kbit / s.
- a CELP coder other than that used in AMR-WB, for example the CELP coder in G.718 to 8. kbit / s.
- other encoders in wide band or operating at frequencies higher than 16 kHz in which the coding of the low band operates at an internal frequency at 12.8 kHz could be used.
- the invention can be obviously adapted to other sampling frequencies 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 a linear prediction, it does not have an excitation signal to be extended, in this case it will be possible to carry out an LPC analysis of the reconstructed signal in the current frame and calculate an LPC excitation. so as to be able to apply the invention.
- the excitation (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 is then calculated over a greater length and the resampling is not carried out in the field of the transform.
- all the calculations necessary for the estimation of the gains ⁇ G HBN , g HB1 (m), g HB2 (m), g HBN , ...) can be carried out in a field logarithmic.
- the combination block 704 can be realized in different ways.
- a privileged way we consider an adaptive additive mix by subframe of the form:
- the block 704 realizes the equivalent of the block 101 of FIG. 1.
- the calculation of the factor obliges to calculate the transform of the decoded excitation signal (or the decoded signal itself according to the domain of FIG. calculation of the relative level of noise or flatness spectral flatness "spectral flatness" in English) in low band if this calculation rests on the spectral flatness; in variants, including the use of a linear regression described above, such a transform is not necessary.
- An exemplary embodiment of such type FIR adaptive bandpass filtering is given in the tables below defining the impulse response of the FIR filter according to the bit rate.
- the scaling step (E407 in FIG. 4) is performed by blocks 508 and 509 identical to FIG.
- the filtering step (E404 of FIG. 4) is performed by the filtering module (block 510) identical to that described with reference to FIG. 5.
- the low band excitation u (n) and the LPC 1 / ⁇ filter (z) will be estimated per frame, by LPC analysis of a low band signal whose band must be extended.
- the low band excitation signal is then extracted by analyzing the audio signal.
- the low band audio signal is resampled before the excitation extraction step, so that the excitation extracted from the audio signal (by linear prediction) is already resolved. sampled.
- the invention illustrated in FIG. 5, or alternatively in FIG. 7, applies in this case to a low band which is not decoded but analyzed.
- FIG. 8 represents an exemplary hardware embodiment of a band extension device 800 according to the invention. This may be an integral part of an audio-frequency signal decoder or equipment receiving decoded or non-decoded audio signals.
- This type of device comprises a PROC processor cooperating with a memory block
- BM having a memory storage and / or working MEM.
- Such a device comprises an input module E able to receive a decoded or extracted excitation audio signal in a first so-called low band frequency band (u (n) or U (k)) and the parameters of a filter of linear prediction synthesis (A (z)). It comprises an output module S adapted to transmit the synthesized high frequency signal (HF_syn) for example to a delay application module such as block 310 of FIG. 3 or to a resampling module such as module 311 .
- a delay application module such as block 310 of FIG. 3
- a resampling module such as module 311 .
- the memory block may advantageously comprise a computer program comprising code instructions for implementing the steps of the band extension method in the sense of the invention, when these instructions are executed by the processor PROC, and in particular the steps for obtaining an extended signal in at least a second frequency band greater than the first frequency band from an oversampled and extended excitation signal in at least a second frequency band, scaling of the signal extended by a subframe defined gain based on a frame and subframe energy ratio and filtering said extended signal scaled by a linear prediction filter whose coefficients are derived from the coefficients of the low band filter.
- the description of FIG. 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 thereof.
- 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 FIG. 3 in addition to the band extension functions according to the invention.
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FR1356100A FR3007563A1 (fr) | 2013-06-25 | 2013-06-25 | Extension amelioree de bande de frequence dans un decodeur de signaux audiofrequences |
PCT/FR2014/051563 WO2014207362A1 (fr) | 2013-06-25 | 2014-06-24 | Extension améliorée de bande de fréquence dans un décodeur de signaux audiofréquences |
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EP (1) | EP3014611B1 (fr) |
CN (1) | CN105324814B (fr) |
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CA3045686C (fr) | 2010-04-09 | 2020-07-14 | Dolby International Ab | Melangeur elevateur audio fonctionnel en mode de prediction ou de non-prediction |
EP3182411A1 (fr) | 2015-12-14 | 2017-06-21 | Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. | Appareil et procédé de traitement de signal audio codé |
US10249307B2 (en) | 2016-06-27 | 2019-04-02 | Qualcomm Incorporated | Audio decoding using intermediate sampling rate |
EP3382702A1 (fr) * | 2017-03-31 | 2018-10-03 | Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. | Appareil et procédé permettant de déterminer une caractéristique prédéterminée liée à un traitement de limitation de bande passante artificielle d'un signal audio |
US10825467B2 (en) * | 2017-04-21 | 2020-11-03 | Qualcomm Incorporated | Non-harmonic speech detection and bandwidth extension in a multi-source environment |
US20190051286A1 (en) * | 2017-08-14 | 2019-02-14 | Microsoft Technology Licensing, Llc | Normalization of high band signals in network telephony communications |
CN107886966A (zh) * | 2017-10-30 | 2018-04-06 | 捷开通讯(深圳)有限公司 | 终端及其优化语音命令的方法、存储装置 |
EP3553777B1 (fr) * | 2018-04-09 | 2022-07-20 | Dolby Laboratories Licensing Corporation | Dissimulation de perte de paquets à faible complexité pour des signaux audio transcodés |
CN110660409A (zh) * | 2018-06-29 | 2020-01-07 | 华为技术有限公司 | 一种扩频的方法及装置 |
CN110556122B (zh) * | 2019-09-18 | 2024-01-19 | 腾讯科技(深圳)有限公司 | 频带扩展方法、装置、电子设备及计算机可读存储介质 |
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DE10041512B4 (de) * | 2000-08-24 | 2005-05-04 | Infineon Technologies Ag | Verfahren und Vorrichtung zur künstlichen Erweiterung der Bandbreite von Sprachsignalen |
US6889182B2 (en) * | 2001-01-12 | 2005-05-03 | Telefonaktiebolaget L M Ericsson (Publ) | Speech bandwidth extension |
SE522553C2 (sv) * | 2001-04-23 | 2004-02-17 | Ericsson Telefon Ab L M | Bandbreddsutsträckning av akustiska signaler |
US6988066B2 (en) * | 2001-10-04 | 2006-01-17 | At&T Corp. | Method of bandwidth extension for narrow-band speech |
ATE331280T1 (de) * | 2001-11-23 | 2006-07-15 | Koninkl Philips Electronics Nv | Bandbreitenvergrösserung für audiosignale |
AU2003260958A1 (en) * | 2002-09-19 | 2004-04-08 | Matsushita Electric Industrial Co., Ltd. | Audio decoding apparatus and method |
US20050004793A1 (en) * | 2003-07-03 | 2005-01-06 | Pasi Ojala | Signal adaptation for higher band coding in a codec utilizing band split coding |
KR100707174B1 (ko) * | 2004-12-31 | 2007-04-13 | 삼성전자주식회사 | 광대역 음성 부호화 및 복호화 시스템에서 고대역 음성부호화 및 복호화 장치와 그 방법 |
CA2603246C (fr) * | 2005-04-01 | 2012-07-17 | Qualcomm Incorporated | Systemes, procedes et appareil de filtrage anti-dispersion |
KR101171098B1 (ko) * | 2005-07-22 | 2012-08-20 | 삼성전자주식회사 | 혼합 구조의 스케일러블 음성 부호화 방법 및 장치 |
US9454974B2 (en) * | 2006-07-31 | 2016-09-27 | Qualcomm Incorporated | Systems, methods, and apparatus for gain factor limiting |
US20090201983A1 (en) * | 2008-02-07 | 2009-08-13 | Motorola, Inc. | Method and apparatus for estimating high-band energy in a bandwidth extension system |
WO2010028297A1 (fr) * | 2008-09-06 | 2010-03-11 | GH Innovation, Inc. | Extension sélective de bande passante |
WO2010036061A2 (fr) * | 2008-09-25 | 2010-04-01 | Lg Electronics Inc. | Appareil pour traiter un signal audio et procédé associé |
US8463599B2 (en) * | 2009-02-04 | 2013-06-11 | Motorola Mobility Llc | Bandwidth extension method and apparatus for a modified discrete cosine transform audio coder |
FR2947945A1 (fr) * | 2009-07-07 | 2011-01-14 | France Telecom | Allocation de bits dans un codage/decodage d'amelioration d'un codage/decodage hierarchique de signaux audionumeriques |
CA2780971A1 (fr) * | 2009-11-19 | 2011-05-26 | Telefonaktiebolaget L M Ericsson (Publ) | Extension de largeur de bande de signal d'excitation ameliore |
US8600737B2 (en) * | 2010-06-01 | 2013-12-03 | Qualcomm Incorporated | Systems, methods, apparatus, and computer program products for wideband speech coding |
MX2013009295A (es) * | 2011-02-15 | 2013-10-08 | Voiceage Corp | Dispositivo y método para cuantificar ganancias de contribuciones adaptativas y fijas de una excitación en un codec celp. |
US20140019125A1 (en) * | 2011-03-31 | 2014-01-16 | Nokia Corporation | Low band bandwidth extended |
WO2013066238A2 (fr) * | 2011-11-02 | 2013-05-10 | Telefonaktiebolaget L M Ericsson (Publ) | Génération d'une extension à bande haute d'un signal audio à bande passante étendue |
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