WO2004097795A2 - Amelioration vocale adaptatvie pour codage audio a faible debit binaire - Google Patents

Amelioration vocale adaptatvie pour codage audio a faible debit binaire Download PDF

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Publication number
WO2004097795A2
WO2004097795A2 PCT/EP2004/004608 EP2004004608W WO2004097795A2 WO 2004097795 A2 WO2004097795 A2 WO 2004097795A2 EP 2004004608 W EP2004004608 W EP 2004004608W WO 2004097795 A2 WO2004097795 A2 WO 2004097795A2
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Prior art keywords
audio signal
signal
accordance
control parameter
processing
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WO2004097795A3 (fr
Inventor
Heiko Purnhagen
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Coding Technologies Sweden AB
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Coding Technologies Sweden AB
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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
    • G10L19/26—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
    • G10L21/0316—Speech enhancement, e.g. noise reduction or echo cancellation by changing the amplitude
    • G10L21/0364—Speech enhancement, e.g. noise reduction or echo cancellation by changing the amplitude for improving intelligibility
    • 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
    • G10L25/93—Discriminating between voiced and unvoiced parts of speech signals

Definitions

  • the present invention relates to low bit rate audio source coding systems and in particular to postprocessing concepts for improving coded and again decoded audio signals.
  • Postprocessing methods in time domain based speech coding sys- terns are disclosed in Chen, Gersho: "Adaptive Postfiltering for Quality Enhancement of Coded Speech," in IEEE Trans. SAP, vol. 3, no 1, pp. 59—71, Jan. 1995.
  • This paper discloses an adaptive postfiltering algorithm for enhancing the perceptual quality of coded speech.
  • the postfilter consists of a long- term postfilter section in cascade with a short-term postfil- ter section and includes spectral tilt compensation and automatic gain control.
  • the long-term section emphasizes pitch harmonics and attenuates the spectral valleys between pitch harmonics.
  • the short-term section emphasizes speech formants and attenuates the spectral valleys between formants. Both filter sections have poles and zeros. Variations of this postfilter are used in several coding standards.
  • this prior art postfiltering method is particularly designed for speech signals, i.e., for signals having a prominent pulse train and, naturally, for signals generated by a time domain based coder/decoder system such as the CELP codec.
  • This object is achieved by a postprocessor in accordance with claim 1, a method of postprocessing in accordance with claim 20, an audio signal encoder in accordance with claim 21, a method for audio encoding in accordance with claim 28 or a com- puter program in accordance with claim 29.
  • the present invention is based on the finding that postprocessing for signal parts which have a specified characteristic which is improved by a postprocessing filter has to be applied to an audio signal only, when the audio signal has this specified characteristic.
  • this fact is to be signaled to a controller using a control parameter.
  • the con- troller is adapted to control the processing filter so that the processing filter is active for the portion of the audio signal, which is indicated as having the specified characteristic, while, in accordance with the present invention, the processing filter is inactive for a portion of the audio signal which does not have the specified characteristic.
  • the inventive concept lies in an adaptive postprocessing in that only parts of the audio signal which take benefit from postprocessing are filtered, while other portions of the audio signal, which do not profit from postprocessing but have a high probability to be distorted are not filtered by means of the processing filter. For these parts, the processing filter is bypassed.
  • the specified characteristic is a voice characteristic
  • the processing filter is a postfilter which is based on a long term prediction.
  • a control parameter is a voicedness parameter, which is preferably calculated by a correlation procedure to determine the degree of voicedness of an examined signal portion.
  • Signal analysis parameters suitable for being input into this filter are the pitch of the voice signal or the pitch lag, which is the reciprocal value of the pitch.
  • the present invention introduces adaptive processing techniques to improve the perceived quality of voice signals like human speech conveyed by means of named coding systems and thereby increase the coding efficiency of the combined system.
  • This adaptive processing is especially suited for general audio codecs, like transform-based codecs, and can also be employed in general audio codecs that utilize high frequency reconstruction (HFR) techniques.
  • HFR high frequency reconstruction
  • adaptive processing of the signal preferably in the decoder is employed in order to reduce such artefacts.
  • This processing is primarily applied to the voiced parts of the audio signal.
  • the adaptive processing attenuates the coding distortion added by the general audio coder while leaving the actual speech signal unmodified.
  • the adaptive processing preferably tracks closely the characteristics of the speech signal, like for example the fundamental frequency, or pitch lag.
  • the adaptive processing can be implemented by a time-variant filter.
  • An aspect of the invention is a method or apparatus for voice enhancement by adaptive processing in an audio codec, where, at an encoder, control parameters are calculated by signal analysis of an input signal, and, at a decoder, an enhanced output signal is generated, using said parameters conveyed from an encoder to control an adaptive processing.
  • a further aspect of the invention is a method or apparatus for voice enhancement by adaptive processing in an audio codec, where at a decoder, control parameters are calculated by sig- nal analysis of a decoded signal and an enhanced output signal is generated, using the parameters to control an adaptive processing.
  • the adaptive processing can be carried out as post-processing for a general audio decoder or the adaptive processing can be carried out as intermediate processing in a general audio decoder enhanced by high frequency reconstruction (HFR) , and as such applied between the core general audio decoder and the HFR processing.
  • HFR high frequency reconstruction
  • Absolute or time differential quantization and entropy coding techniques are employed to convey control parameters from an encoder to a decoder as side information in a bit stream.
  • voice enhancement processing is, adaptively, only applied if beneficial for subjective quality of the signal generated by a decoder, e.g. if the signal processed by the codec system is detected to be human speech.
  • the voice enhancement processing is preferably carried out by pitch lag and pitch intensity adaptive FIR or IIR filtering.
  • Processing parameters like pitch lag and extent or degree of processing are updated once or several times per time frame of an general audio codec.
  • Filter parameters used in enhancement processing are preferably continuously interpolated based on and between parameter updates calculated by signal analysis in a decoder or conveyed from an encoder.
  • Adaptive selection of interpolation mode (cross-fade vs. sweep) is done depending on properties of the processed audio signal.
  • Voice enhancement processing is preferably only applied to certain frequency regions of the output signal of a general audio coder while the remaining frequency regions remain un- processed. These frequency regions can be low pass regions, in which the most part of the voice spectrum is included.
  • Fig. 1 illustrates the adaptive processing for voice enhancement in an audio decoder with signal analysis in the decoder
  • Fig. 2 illustrates an audio encoder with additional signal analysis and optional speech detection generating side information conveyed in the bit stream for control of adaptive processing for voice enhancement in a decoder;
  • Fig. 3 illustrates the " adaptive processing for voice enhancement in an audio decoder controlled by side information conveyed from the encoder
  • Fig. 4 illustrates the impulse response of an time-variant adaptive FIR filter employed in the audio decoder for voice enhancement
  • Fig. 5 illustrates a means for applying voice enhancement according to the present invention
  • Fig. 6 illustrates a means for providing enhancement control parameters according to the present invention
  • Fig. 7 illustrates an inventive decoder apparatus for adaptive voice enhancement for low bit rate audio coding
  • Fig. 8 illustrates a means for voice enhancement processing, according to the present invention
  • Fig. 9 illustrates an inventive encoder apparatus for adaptive voice enhancement for low bit rate audio coding
  • Fig. 10 illustrates pseudo code for a temporal parameter extrapolation method
  • Fig. 11 illustrates pseudo code for a time-variant filtering method with two alternative parameter interpolation methods.
  • General audio codecs are commonly based on a time/frequency decomposition.
  • the audio signal is processed as a sequence of short, overlapping signal segments, or frames.
  • a win- dowed signal segment is transformed into a frequency domain, using a transform like an MDCT, and the frequency domain representation of the segment is quantized and coded.
  • the operation of the quantizer is controlled by a psychoacoustic model in order to adapt it to the perceptual properties of the pre- sent audio signal.
  • the frequency domain representation of the signal segments is reconstructed, transformed back to the time domain, and windowed.
  • Overlap-add is used to obtain the output signal of the decoder.
  • HFR high frequency reconstruction
  • adaptive processing of the signal in the decoder is employed in order to reduce such artefacts.
  • This processing is primarily applied to the voiced parts of the audio signal.
  • the adaptive processing attenuates the coding distortion added by the general audio coder while leaving the actual speech signal unmodified.
  • the adaptive processing has to track closely the characteristics of the speech signal, like for example the fundamental frequency, or pitch lag.
  • the extent of processing has to be adapted as to the signal characteristics as well. In this way, the processing is disabled adaptively if it would have a negative effect on the perceived quality, for example, if the present signal is not a speech signal.
  • the adaptive processing can be implemented by a time-variant filter.
  • Voice enhancement processing can either be done by extending only the audio decoder by modules for voice signal analysis, parameter interpolation, and voice enhancement processing, as shown in Fig. 1. In this case, no side information is conveyed from the encoder and no extension of the encoder is needed.
  • the encoder can be extended by the voice signal analysis module and the control parameters for the enhancement process calculated in this module are appropriately quantized and coded and conveyed as side information in the bit stream, as shown in Fig. 2.
  • the decoder is only extended by modules for parameter interpolation and voice enhancement processing, as shown in Fig. 3.
  • This alternative has the advantage that computational complexity in the decoder is re- prised and that a clean, uncoded signal is available for signal analysis in the encoder.
  • the disadvantage is that some bit rate is needed to convey " the control parameters from the encoder to the decoder in the bit stream.
  • the presented system with adaptive voice enhancement is intended for used with arbitrary audio signal, e.g. speech, music, other sounds, or combination thereof.
  • the voice signal analysis module can optionally be augmented by a speech detection module, as also shown in Fig. 2, in order to adap- tively activate the voice enhancement processing only if this is beneficial for the perceived quality of the signal generated in the decoder, which, for example, can be the case if the signal processed by the codec is detected to be a human speech signal.
  • Signal analysis can be done by calculating and processing the short time autocorrelation function of the audio signal or a sample-rate reduced or bandwidth limited version thereof.
  • Other techniques to analyse the signal for periodicity and to estimate the intensity (e.g. voicedness) and periodicity- interval (e.g. a pitch lag) of the audio signal can also be used.
  • a short segment of the signal for example with a length of approximately 20 ms
  • the difference in temporal location of both segments is an estimate of the periodicity interval of the analysed signal.
  • this periodicity interval is also re- ferred to as "pitch lag" and corresponds to the reciprocal value of the fundamental frequency.
  • the value of the normal- ' ized cross-correlation gives information about the pitchedness of the analysed signal segment, where 1 indicates a strongly pitched signal and 0 indicates an unpitched signal. Integer multiples of the original pitch lag also exhibit large cross- correlation, so that this ambiguity has to be taken care of when estimating pitch lag.
  • tracking mechanisms that take into account earlier estimates can be em- ployed.
  • the voice enhancement process is carried out by adaptively filtering the output signal of the general audio decoder.
  • the adap- tive filter is operated such that it leaves the spectral components of a voiced signal unchanged while attenuating the artefacts introduced by the general audio coding process at frequencies different from those of the spectral components of the voiced signal.
  • the adaptive filter which can be imple- mented as a time-variant FIR or IIR filter, has to follow the time-varying pitch lag estimated for the signal being processed very precisely.
  • the intensity of processing should be reduced when the signal is not fully pitched, and processing should be completely disabled when the pitched- ness falls below an appropriate threshold. This can be achieved by an appropriate function mapping the observed pitchedness to the processing intensity c in the range 0 to 1 as defined below.
  • the enhancement processing parameters have to be calculated on a sufficiently fine time grid. This means that typically several subsequent processing parameters sets are calculated for one audio frame of the general audio codec, with which the present enhancement processing is combined. For example, if the frame length of the general audio coder is approximately 40 s, the processing parameters can be updated approximately every 10 ms or 20 ms. A fixed or adaptive time grid can be used in the case where these parameters are conveyed from the encoder to the decoder.
  • enhancement control parameters namely pitch lag parameters d( ) and processing intensity parameters c ⁇ t
  • appropriate quantization and coding for example employing Huffman codes
  • both pitch lag and processing intensity often vary only slowly over time, it can be advantageous to provide means for both absolute and time-differential coding of these parameters. In this case it is also necessary to signal which of these both coding methods was employed for each parameter of groups of parameters.
  • interpolation of the pitch lag parameter with sub-sample accuracy is employed to achieve a smoothly time-variant filter.
  • the processing intensity parameter is interpolated between two subsequent parameter sets.
  • cross-fade of the impulse response between two subsequent pitch lag / processing intensity parameters can employed.
  • the pitch estimate is valid for the center of the current signal segment, i.e., the signal segment which is being compared to earlier signal " segments .
  • the most current pitch estimate is valid for a point of time that is 10 ms apart from (i.e., "older") the most recent
  • the pitchedness or processing intensity can be extrapolated in a similar way. After extrapolation, it is made sure that the extrapolated parameters stay within their respective permissible range.
  • the pseudo code fragment shown in Fig. 10 illustrates a possible extrapolation technique as outlined above.
  • the following sweep interpolation of the time-variant filter can be em- ployed.
  • the parameters of the enhancement filter are correspondingly adapted.
  • H ⁇ Z ,t) b 0 ⁇ t) + b la (t).z- ⁇ + b lb ⁇ t)-z- ⁇ this can, for example, be achieved by ensuring
  • the pitch lag d ⁇ t can be interpolated between two given values d ⁇ T x ) and d ⁇ T 2 ) as
  • processing intensity c(t) can be inter- polated between two given values c ⁇ T x ) and c ⁇ T 2 ) as
  • Fig. 4 shows the impulse response of a time-variant FIR filter at time t .
  • d ⁇ t denotes the current pitch lag.
  • this pitch lag has a non-integer value.
  • 6, is split up into ⁇ l ⁇ and b lb hich are applied to the two integer values k and k + l that are nearest to the desired pitch lag d ⁇ t) .
  • the following cross-fade interpolation of the time-variant filter can be employed.
  • the parameters of the enhancement filter are correspondingly adapted.
  • an FIR filter based enhancement processing with the time variant impulse response h(n,t) at time t having the following transfer function
  • the selection of sweep vs. cross-fade interpolation method is based on the relative change of the pitch lag. If this change exceeds one or more given thresholds, cross-fade interpolation is used instead of sweep interpolation.
  • the selection of the interpolation method can be based on the bit that signals whether absolute or time-differential coding is employed. For time-differential coding, sweep interpolation is used while for absolute coding, cross-fade interpolation is used.
  • Fig. 7 illustrates an inventive decoder apparatus for adaptive voice enhancement for low bit rate audio coding. In this particular exemplification of the present invention the adaptive voice enhancement is operated on the decoded signal from a general audio decoder, but prior to the high frequency reconstruction module. The decoder apparatus operates on a bit stream being transmit to the decoder or read from file.
  • the bit stream corresponding to the general audio decoder is input to the means for general audio decoding 701.
  • the means for general audio decoding can be an arbitrary wave-form coder, ' such as MPEG-2/4 AAC.
  • The" output from the means for general audio decoding 701 is input to the means for voice enhancement processing 702.
  • the output from the means for general audio decoding is in this particular exemplification of the present invention a time-domain signal.
  • the output from the means for voice enhancement processing 702 is input to the means for HFR (High Frequency Reconstruction) processing 703.
  • the means for HFR processing can be an arbitrary HFR method such as MPEG-4 SBR.
  • the means for HFR processing are optional and that the apparatus for voice enhancement for low bit rate audio coding can be operated without the means for HFR.
  • means for HFR are included in the inventive apparatus, it is an essential feature of the present invention to use the voice enhancement processing between the general audio decoder and the high frequency reconstruction.
  • Fig. 8. illustrates the means for voice enhancement processing 702 in more detail.
  • the output from the means for general audio decoding 701 in Fig. 7, is input to the means for applying voice enhancement processing 802.
  • Fig. 6 illustrates a more detailed description of the means for providing enhancement control parameters.
  • the inventive apparatus for providing enhancement control parameters can be part of an encoder apparatus that incorporates the control pa- rameters into the bit stream.
  • the apparatus for providing enhancement control parameters can be part of the decoder if the control parameters are not part of the bit stream and therefore need to be estimated in the decoder apparatus.
  • the audio signal being the either the original input signal if the apparatus is used as part of an encoder apparatus, or being the decoded output signal of the means for general audio decoding 701 in the decoder apparatus, is input to a means for estimation of the pitch lag and voicedness, 601, as well as to a means for detection of speech 604.
  • the output from the means for estimation of pitch lag and voicedness 601 is input to a means for calculating enhancement control parameters 602.
  • the output from the means for detection of speech is also input to 602.
  • the means 602 is operative to estimate a long term prediction filter, based on the pitch-lag and voicedness of the audio signal as well as on a the outcome from the means for detection of speech.
  • the output from the means for calculating enhancement control parameters 602, is input to a means for temporal parameter extrapolation, 603. It is important to note that 603 and 604 are optional and the in- ventive apparatus for providing control parameters can be operated without them.
  • Fig.5 illustrates in more detail the inventive means for applying voice enhancement processing 802 in Fig. 8.
  • the en- hancement control parameters 506 are input to a means for cross fade interpolation 502, as well as to a means for sweep interpolation 503, and to a means for selection of interpolation method 504.
  • the output from 504 is used to select 505 which output, filter coefficient from means for cross-fade interpolation 502, or filter coefficients from means for sweep interpolation 503, is input to a means for time-variant filtering 501.
  • the audio signal output from means for general audio decoding 701 is input to the means for time-variant fil- tering 501.
  • Fig. 9. illustrates an inventive encoder apparatus for adaptive voice enhancement for low bit rate audio coding.
  • the original input audio signal is input to a means for providing enhancement control parameters 901, and to a means for general audio encoding 902.
  • the means for general audio encoding can be a wave-form coder such as MPEG-2/4 AAC, and can also include an HFR encoder such as MPEG-4 SBR.
  • the means for providing enhancement control parameters are similar to the means for providing enhancement control parameters 801, in the inventive decoder apparatus, when the means ⁇ Oloperate on the output from audio decoding.
  • the output from the means for providing enhancement control parameters 901, is input to a means for quantization and encoding 903.
  • the output from 903 and 902 constitutes the bit stream data being sent to a decoder apparatus or stored in a file for subsequent decoding by a decoder apparatus .
  • a pre-processed version of this signal can be used as input to a means for general audio encoding 902.
  • the means for pre-processing can utilized the provided enhancement control parameters in order to compensate for undesired effects that can occur when the means for applying voice enhancement processing is operated in the decoder.
  • the inventive device can be implemented in hardware or in software or in a firmware including hardware constituents and software constituents.
  • the invention also is a computer program having a computer-readable code for carrying out the inventive methods when running on a computer.

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  • Engineering & Computer Science (AREA)
  • Computational Linguistics (AREA)
  • Signal Processing (AREA)
  • Health & Medical Sciences (AREA)
  • Audiology, Speech & Language Pathology (AREA)
  • Human Computer Interaction (AREA)
  • Physics & Mathematics (AREA)
  • Acoustics & Sound (AREA)
  • Multimedia (AREA)
  • Compression, Expansion, Code Conversion, And Decoders (AREA)
  • Quality & Reliability (AREA)

Abstract

La présente invention concerne des améliorations à des codeurs-décodeurs de fabrication actuelle appliqués à des signaux vocaux tels que la parole. Ces améliorations sont rendues possibles par un traitement adaptatif (501) du signal dans le décodeur, qui réduit les distorsions perçues introduites dans le processus de codage, principalement pour les parties vocales du signal audio. Le traitement du signal est commandé de manière adaptative par l'analyse de ce signal dans le codeur ou dans le décodeur. L'invention peut être appliquée en tant que post-traitement dans un codeur-décodeur audio général ou comme opération de traitement intermédiaire dans un codeur-décodeur audio général faisant intervenir des techniques de reconstruction haute fréquence.(HFR).
PCT/EP2004/004608 2003-04-30 2004-04-30 Amelioration vocale adaptatvie pour codage audio a faible debit binaire Ceased WO2004097795A2 (fr)

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SE0301272A SE0301272D0 (sv) 2003-04-30 2003-04-30 Adaptive voice enhancement for low bit rate audio coding

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Cited By (2)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US8818541B2 (en) 2009-01-16 2014-08-26 Dolby International Ab Cross product enhanced harmonic transposition
EP2492911A4 (fr) * 2009-10-21 2015-04-15 Panasonic Ip Man Co Ltd Appareil d'encodage audio, appareil de décodage, procédé, circuit et programme

Family Cites Families (3)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
JP3653826B2 (ja) * 1995-10-26 2005-06-02 ソニー株式会社 音声復号化方法及び装置
DE10124421C1 (de) * 2001-05-18 2002-10-17 Siemens Ag Verfahren zur Schätzung eines Codecparameters
WO2003036621A1 (fr) * 2001-10-22 2003-05-01 Motorola, Inc., A Corporation Of The State Of Delaware Procede et appareil permettant d'ameliorer la sonie d'un signal audio

Cited By (10)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US8818541B2 (en) 2009-01-16 2014-08-26 Dolby International Ab Cross product enhanced harmonic transposition
US9799346B2 (en) 2009-01-16 2017-10-24 Dolby International Ab Cross product enhanced harmonic transposition
US10192565B2 (en) 2009-01-16 2019-01-29 Dolby International Ab Cross product enhanced harmonic transposition
US10586550B2 (en) 2009-01-16 2020-03-10 Dolby International Ab Cross product enhanced harmonic transposition
US11031025B2 (en) 2009-01-16 2021-06-08 Dolby International Ab Cross product enhanced harmonic transposition
US11682410B2 (en) 2009-01-16 2023-06-20 Dolby International Ab Cross product enhanced harmonic transposition
US11935551B2 (en) 2009-01-16 2024-03-19 Dolby International Ab Cross product enhanced harmonic transposition
US12119011B2 (en) 2009-01-16 2024-10-15 Dolby International Ab Cross product enhanced harmonic transposition
US12165666B2 (en) 2009-01-16 2024-12-10 Dolby International Ab Cross product enhanced harmonic transposition
EP2492911A4 (fr) * 2009-10-21 2015-04-15 Panasonic Ip Man Co Ltd Appareil d'encodage audio, appareil de décodage, procédé, circuit et programme

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SE0301272D0 (sv) 2003-04-30

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