EP1383110A1 - Verfahren und Vorrichtung für Breitbandsprachkodierung, insbesondere mit einer verbesserten Qualität der stimmhaften Rahmen - Google Patents
Verfahren und Vorrichtung für Breitbandsprachkodierung, insbesondere mit einer verbesserten Qualität der stimmhaften Rahmen Download PDFInfo
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- EP1383110A1 EP1383110A1 EP02015920A EP02015920A EP1383110A1 EP 1383110 A1 EP1383110 A1 EP 1383110A1 EP 02015920 A EP02015920 A EP 02015920A EP 02015920 A EP02015920 A EP 02015920A EP 1383110 A1 EP1383110 A1 EP 1383110A1
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Images
Classifications
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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
Definitions
- the invention relates to speech encoding / decoding extended band, in particular but not limited to telephony mobile.
- the bandwidth of the speech signal is between 50 and 7000 Hz.
- Successive speech sequences sampled at one predetermined sampling frequency are processed in a coding device using a prediction linear excitation by coded sequences (ACELP: “algebraic-code-excited linear-prediction ”), well known to those skilled in the art, and described in particular in recommendation ITU-TG 729, version 3/96, titled “speech coding at 8 kbit / s by prediction linear with excitation by coded sequences with algebraic structure conjugate ”.
- ACELP “algebraic-code-excited linear-prediction ”
- the prediction coder CD of the ACELP type, is based on the linear predictive coding model with code excitation.
- the coder operates on vocal superframes equivalent for example to 20 ms of signal and each comprising 320 samples.
- the extraction of the linear prediction parameters i.e. the coefficients of the linear prediction filter, also called short-term synthesis filter 1 / A (z), is carried out for each speech superframe.
- each superframe is subdivided into 5 ms frames comprising 80 samples.
- the speech signal is analyzed to extract the parameters of the CELP prediction model (that is to say, in particular, a long-term digital excitation word v i extracted from an adaptive coded DLT directory, also called “adaptive long term dictionary", an associated long term gain Ga, a short term excitation word c j , extracted from a DCT algebraic coded repertoire, also called “fixed coded repertoire” or “short term dictionary algebraic ", and an associated short-term gain Gc).
- a long-term digital excitation word v i extracted from an adaptive coded DLT directory, also called “adaptive long term dictionary", an associated long term gain Ga
- a short term excitation word c j extracted from a DCT algebraic coded repertoire, also called “fixed coded repertoire” or “short term dictionary algebraic ", and an associated short-term gain Gc).
- these parameters are used, in a decoder, to retrieve the excitation and predictive filter parameters. We then reconstitutes speech by filtering this excitation flow in a short-term synthesis filter.
- the short-term dictionary DCT is founded on an algebraic structure using a permutation model intertwined with Dirac pulses.
- this coded directory which contains innovative excitations also called excitations algebraic or short-term, each vector contains a certain number of non-zero pulses, for example four, each of which can have amplitude +1 or -1 with predetermined positions.
- the CD encoder processing means include functionally of the first MEXT 1 extraction means intended to extract the word long-term excitement, and second MEXT2 extraction means intended to extract the word short-term excitement. Functionally, these means are made for example in software within a processor.
- These extraction means include a predictive filter FP having a transfer function equal to 1 / A (z), as well as a filter FPP perceptual weighting with a transfer function W (z).
- the perceptual weighting filter is applied to the signal to model the perception of the ear.
- the extraction means include means MECM intended to perform a minimization of a square error average.
- the linear prediction FP synthesis filter models the spectral envelope of the signal. Linear predictive analysis is performed all superframes, so as to determine the linear predictive filter coefficients. These are converted to spectral line pairs (LSP: “Line Spectrum Pairs”) and digitized by predictive vector quantization in two stages.
- LSP Line Spectrum Pairs
- Each 20 ms speech superframe is divided into four frames of 5 ms each containing 80 samples.
- the settings Quantized LSPs are transmitted to the decoder once per superframe while long term and short term parameters are passed at each frame.
- the coefficients of the linear prediction filter, quantified and not quantified, are used for the most recent frame of a super-frame, while the other three frames of the same super-frame use an interpolation of these coefficients.
- Tonal delay open loop is estimated every two frames based on the perceptually weighted voice signal. Then, the following operations are repeated at each frame:
- the long-term target signal X LT is calculated by filtering the sampled speech signal s (n) by the perceptual weighting filter FPP.
- the impulse response of the weighted synthesis filter is calculated.
- a closed loop tonal analysis using a minimization of the mean square error is then carried out in order to determine the long-term excitation word v i and the associated gain Ga, by means of the target signal and the impulse response, by searches around the value of the tone delay in open loop.
- the long-term target signal is then updated by subtracting the filtered contribution y from the adaptive coded directory DLT and this new short-term target signal X ST is used when exploring the fixed coded directory DCT in order to determine the password.
- short term excitation C j and the associated gain G c is used when exploring the fixed coded directory DCT in order to determine the password.
- the inventors have in fact observed that even with a very rich algebraic dictionary, the algorithm for encoding the speech produces a reconstructed signal corrupted by different kinds noises and in particular a “whistling” noise which taints voiced speech frames.
- the invention also provides a control type solution gain, but totally different from that described in particular in the articles of Taniguchi and others and of Shoham.
- the invention also aims to control regardless of short-term and long-term distortions.
- the invention therefore provides a speech encoding method with wide band, in which the speech is sampled so as to obtain successive voice frames each comprising a predetermined number of samples, and for each voice frame, we determines parameters of a linear prediction model at excitation by code, these parameters comprising a numeric word of long-term excitement extracted from an adaptive coded repertoire and a associated long-term gain, as well as a word of short-term excitement extract from an algebraic coded repertoire and short-term gain associated, and we update the adaptive coded directory from the word excerpt long term excitement and short term excitement word extract.
- the method includes an update of the state of the linear prediction filter with the word short-term excitation filtered by an order filter greater than or equal to 1, for example an impulse response filter finite of order 1, whose coefficients depend on the value of the gain in the long run, so as to weaken the contribution of excitement to short term when the gain of long term excitement is greater at a predetermined threshold, for example equal to 1.
- the solution according to the invention consists here to weaken the contribution of short-term excitement if the gain of long-term excitement is important.
- this is the contribution of undiminished short-term excitement which is stored in the adaptive dictionary for updating. So the reduction occurs only on exit. Preserving the magnitude of the short-term contribution to be stored is important, since the richness of the adaptive dictionary is thus preserved for the lowest frequencies.
- the gain correction must also be applied when reconstructing the signal at the decoder.
- the first coefficient B0 of the filter is equal to 1 / (1 + ⁇ . min (Ga, 1))
- the second coefficient B1 of the filter is equal to ⁇ .min (Ga, 1) / (1 + ⁇ .min (Ga, 1))
- ⁇ is a real number with a lower absolute value at 1
- Ga is the long-term gain
- min (Ga, 1) designates the minimum value between Ga and 1.
- the extraction of the word of long-term excitement using a first filter of perceptual weighting including a first weighting filter formant we perform the short excitation word extraction term using the first perceptual weighting filter cascaded to a second perceptual weighting filter comprising a second formantic weighting filter.
- the denominator of the transfer function of the first formantic weighting filter equals the numerator of the second weighting filter formant.
- the use of two filters of weighting different formant allows to control regardless of short-term and long-term distortions.
- the short-term weighting filter is cascaded to the filter of long-term weighting.
- tying the denominator of the long-term weighting filter in the numerator of the short-term weighting allows these two to be controlled separately filters and also allows a clear simplification when these two filters are cascaded.
- the first extraction means include a digital filter of linear prediction
- the device comprises second means update capable of updating the filter status of the linear prediction with short term excitation word filtered by a filter of order greater than or equal to 1 whose coefficients depend on the value of the long-term gain, so as to weaken the contribution of short-term excitement when the gain of long-term excitement is above a predetermined threshold.
- the first extraction means include a first weighting filter perceptual with a first weighting filter formantic, by the fact that the second means of extraction include the first perceptual weighting filter cascaded to a second perceptual weighting filter comprising a second formantic weighting filter, and the denominator of the transfer function of the first formantic weighting filter equals the numerator of the second weighting filter formant.
- the invention also relates to a terminal of a system wireless communication, such as a mobile phone cell, incorporating a device as defined above.
- the encoding device, or CD encoder, according to the invention, as illustrated in FIG. 2, differs from that of the prior art as illustrated in FIG. 1 by the fact that the CD encoder further comprises second updating means MAJ2 able to update the state of the linear prediction filter FP and the state of the perceptual weighting filter FPP with the short-term excitation word c j filtered by a filter of order greater than or equal to 1 FLT1 which is for example here a filter of order 1 with finite impulse response.
- second updating means MAJ2 able to update the state of the linear prediction filter FP and the state of the perceptual weighting filter FPP with the short-term excitation word c j filtered by a filter of order greater than or equal to 1 FLT1 which is for example here a filter of order 1 with finite impulse response.
- the coefficients of this first order filter depend on the value long-term gain Ga, so as to weaken the contribution of short-term excitement when gaining long-term excitement Ga is greater than a predetermined threshold, for example equal to 1.
- the transfer function of the filter FLT1 is equal to B0 + B1 z -1 and the first coefficient of the filter B0 can be determined by formula (I) below. 1 / (1 + 0.98 min (Ga, 1)) while the second coefficient of filter B1 can be determined by formula (II) below. 0.98 min (Ga, 1) / (1 + 0.98 min (Ga, 1))
- the filtering of the excitation must also be applied for updating the memory status of the filters in the DCD decoder, as shown diagrammatically in FIG. 2a.
- the variant embodiment illustrated in FIG. 2 allows eliminating hissing type noise on voiced speech frames.
- the FPP perceptual weighting filter uses the masking properties of the human ear compared to the spectral envelope of the speech signal, whose shape is a function resonances of the vocal tract. This filter allows you to assign more importance of the error appearing in the spectral valleys by compared to formic peaks.
- the same FPP perceptual weighting filter is used for short-term research and for long-term research.
- the transfer function W (z) of this FPP filter is given by the formula (III) below.
- W ( z ) AT ( z / ⁇ 1 ) AT ( z / ⁇ 2 ) in which 1 / A (z) is the transfer function of the predictive filter FP and ⁇ 1 and ⁇ 2 are the perceptual weighting coefficients, the two coefficients being positive or zero and less than or equal to 1 with the coefficient ⁇ 2 less than or equal to the coefficient ⁇ 1.
- the perceptual weighting filter consists of a formantic weighting filter and a weighting of the slope of the spectral envelope of the signal (tilt).
- FIG. 3 Such an embodiment is illustrated in FIG. 3, in which, compared to Figure 2, the unique FPP filter was replaced by a first formantic weighting filter FPP1 for long-term research, cascaded with a second filter of FPP2 formant weighting for short-term research.
- the filters appearing in the long-term research loop should also appear in the short-term research loop.
- the transfer function W 1 (z) of the formantic weighting filter FPP1 is given by formula (IV) below.
- W 1 ( z ) AT ( z / ⁇ 11 ) AT ( z / ⁇ 12 ) while the transfer function W 2 (z) of the formantic weighting filter FPP2 is given by the formula (V) below.
- W 2 ( z ) AT ( z / ⁇ 21 ) AT ( z / ⁇ 22 )
- the coefficient ⁇ 12 is equal to the coefficient ⁇ 21 . This allows a clear simplification when cascading these two filters.
- the filter equivalent to the cascade of these two filters has a transfer function given by the formula (VI) below.
- the synthesis filter FP (having the transfer function 1 / A (z)) followed by the long-term weighting filter FPP1 and the weighting filter FPP2 is then equivalent to the filter whose transfer function is given by formula (VII) below. 1 AT ( z / ⁇ 22 )
- the invention advantageously applies to telephony mobile, and in particular to all remote terminals belonging to a wireless communication system.
- Such a terminal for example a TP mobile telephone, such as that illustrated in FIG. 4, conventionally comprises a antenna connected via a DUP duplexer to a chain reception CHR and a CHT transmission chain.
- a baseband processor BB is connected to the chain respectively of reception CHR and to the chain of transmission CHT by via analog digital ADCs and analog digital DACs.
- the processor BB performs processing in baseband, including DCN channel decoding, followed by DCS source decoding.
- the processor For transmission, the processor performs source coding CCS followed by CCN channel coding.
- the mobile phone incorporates an encoder according to the invention, it is incorporated within the coding means of CCS source, while the decoder is incorporated within the means DCS source decoding.
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- Signal Processing (AREA)
- Health & Medical Sciences (AREA)
- Audiology, Speech & Language Pathology (AREA)
- Human Computer Interaction (AREA)
- Physics & Mathematics (AREA)
- Acoustics & Sound (AREA)
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- Compression, Expansion, Code Conversion, And Decoders (AREA)
Priority Applications (3)
| Application Number | Priority Date | Filing Date | Title |
|---|---|---|---|
| EP02015920A EP1383110A1 (de) | 2002-07-17 | 2002-07-17 | Verfahren und Vorrichtung für Breitbandsprachkodierung, insbesondere mit einer verbesserten Qualität der stimmhaften Rahmen |
| EP03291748A EP1383112A3 (de) | 2002-07-17 | 2003-07-15 | Verfahren und Vorrichtung zur Sprachkodierung mit erhöhter Bandbreite, insbesondere mit einer erhöhten Qualität stimmhafter Sprachrahmen |
| US10/622,020 US20040064312A1 (en) | 2002-07-17 | 2003-07-17 | Method and device for encoding wideband speech, allowing in particular an improvement in the quality of the voiced speech frames |
Applications Claiming Priority (1)
| Application Number | Priority Date | Filing Date | Title |
|---|---|---|---|
| EP02015920A EP1383110A1 (de) | 2002-07-17 | 2002-07-17 | Verfahren und Vorrichtung für Breitbandsprachkodierung, insbesondere mit einer verbesserten Qualität der stimmhaften Rahmen |
Publications (1)
| Publication Number | Publication Date |
|---|---|
| EP1383110A1 true EP1383110A1 (de) | 2004-01-21 |
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Family Applications (1)
| Application Number | Title | Priority Date | Filing Date |
|---|---|---|---|
| EP02015920A Withdrawn EP1383110A1 (de) | 2002-07-17 | 2002-07-17 | Verfahren und Vorrichtung für Breitbandsprachkodierung, insbesondere mit einer verbesserten Qualität der stimmhaften Rahmen |
Country Status (2)
| Country | Link |
|---|---|
| US (1) | US20040064312A1 (de) |
| EP (1) | EP1383110A1 (de) |
Families Citing this family (1)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| DE602004025517D1 (de) * | 2004-05-17 | 2010-03-25 | Nokia Corp | Audiocodierung mit verschiedenen codierungsrahmenlängen |
Citations (3)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| EP0749110A2 (de) * | 1995-06-07 | 1996-12-18 | AT&T IPM Corp. | Adaptives, auf der Grundlage eines Kodebuchs arbeitendes Sprachkompressionssystem |
| WO2002007061A2 (en) * | 2000-07-14 | 2002-01-24 | Conexant Systems, Inc. | A speech communication system and method for handling lost frames |
| US6385573B1 (en) * | 1998-08-24 | 2002-05-07 | Conexant Systems, Inc. | Adaptive tilt compensation for synthesized speech residual |
Family Cites Families (4)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| US5699485A (en) * | 1995-06-07 | 1997-12-16 | Lucent Technologies Inc. | Pitch delay modification during frame erasures |
| US6330533B2 (en) * | 1998-08-24 | 2001-12-11 | Conexant Systems, Inc. | Speech encoder adaptively applying pitch preprocessing with warping of target signal |
| US7606703B2 (en) * | 2000-11-15 | 2009-10-20 | Texas Instruments Incorporated | Layered celp system and method with varying perceptual filter or short-term postfilter strengths |
| US7353168B2 (en) * | 2001-10-03 | 2008-04-01 | Broadcom Corporation | Method and apparatus to eliminate discontinuities in adaptively filtered signals |
-
2002
- 2002-07-17 EP EP02015920A patent/EP1383110A1/de not_active Withdrawn
-
2003
- 2003-07-17 US US10/622,020 patent/US20040064312A1/en not_active Abandoned
Patent Citations (3)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| EP0749110A2 (de) * | 1995-06-07 | 1996-12-18 | AT&T IPM Corp. | Adaptives, auf der Grundlage eines Kodebuchs arbeitendes Sprachkompressionssystem |
| US6385573B1 (en) * | 1998-08-24 | 2002-05-07 | Conexant Systems, Inc. | Adaptive tilt compensation for synthesized speech residual |
| WO2002007061A2 (en) * | 2000-07-14 | 2002-01-24 | Conexant Systems, Inc. | A speech communication system and method for handling lost frames |
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| Publication number | Publication date |
|---|---|
| US20040064312A1 (en) | 2004-04-01 |
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