EP2162883A2 - Limitation de distorsion introduite par un post-traitement au decodage d'un signal numerique - Google Patents
Limitation de distorsion introduite par un post-traitement au decodage d'un signal numeriqueInfo
- Publication number
- EP2162883A2 EP2162883A2 EP08806164A EP08806164A EP2162883A2 EP 2162883 A2 EP2162883 A2 EP 2162883A2 EP 08806164 A EP08806164 A EP 08806164A EP 08806164 A EP08806164 A EP 08806164A EP 2162883 A2 EP2162883 A2 EP 2162883A2
- Authority
- EP
- European Patent Office
- Prior art keywords
- post
- signal
- value
- current amplitude
- decoded
- Prior art date
- Legal status (The legal status is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the status listed.)
- Granted
Links
- 238000012805 post-processing Methods 0.000 title claims abstract description 37
- 238000012545 processing Methods 0.000 claims abstract description 13
- 230000009467 reduction Effects 0.000 claims abstract description 10
- 238000013139 quantization Methods 0.000 claims description 45
- 238000000034 method Methods 0.000 claims description 21
- 230000015654 memory Effects 0.000 claims description 11
- 230000006870 function Effects 0.000 claims description 8
- 238000004590 computer program Methods 0.000 claims description 4
- 230000002123 temporal effect Effects 0.000 claims description 2
- 238000011282 treatment Methods 0.000 description 7
- 230000003111 delayed effect Effects 0.000 description 6
- 238000007906 compression Methods 0.000 description 5
- 230000006835 compression Effects 0.000 description 4
- 238000001914 filtration Methods 0.000 description 4
- 238000011002 quantification Methods 0.000 description 4
- 230000003044 adaptive effect Effects 0.000 description 3
- 230000004044 response Effects 0.000 description 2
- 238000012360 testing method Methods 0.000 description 2
- 230000007423 decrease Effects 0.000 description 1
- 238000005259 measurement Methods 0.000 description 1
- 238000005457 optimization Methods 0.000 description 1
- 230000008447 perception Effects 0.000 description 1
- 230000008569 process Effects 0.000 description 1
- 230000005236 sound signal Effects 0.000 description 1
- 230000001629 suppression Effects 0.000 description 1
- 230000007704 transition Effects 0.000 description 1
- 238000011144 upstream manufacturing Methods 0.000 description 1
- 238000012795 verification Methods 0.000 description 1
- 230000003936 working memory Effects 0.000 description 1
Classifications
-
- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
- G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
- G10L19/04—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using predictive techniques
- G10L19/26—Pre-filtering or post-filtering
-
- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
- G10L21/00—Speech or voice signal processing techniques to produce another audible or non-audible signal, e.g. visual or tactile, in order to modify its quality or its intelligibility
- G10L21/02—Speech enhancement, e.g. noise reduction or echo cancellation
Definitions
- the present invention relates to signal processing, in particular digital signals in the telecommunications field, these signals being, for example, speech, music, video signals, or other signals.
- the rate needed to pass an audio and / or video signal with sufficient quality is an important parameter in telecommunications.
- audio coders have been developed in particular to compress the amount of information necessary to transmit a signal.
- Some encoders achieve particularly high information compression rates. Such coders generally use advanced information modeling and quantification techniques. Thus, these encoders only transmit models or partial data of the signal.
- the decoded signal although it is not identical to the original signal (since part of the information has not been transmitted because of the quantization operation), nevertheless remains very close to the signal of origin (at least from the point of view of perception).
- Quantization noise The difference, in the mathematical sense, between the decoded signal and the original signal is then called "quantization noise".
- Another treatment family is the conventional noise reduction treatments that distinguish the wanted signal from the spurious noises and that can be applied as post-processing to reduce the quantization noise after decoding.
- This type of processing originally makes it possible to reduce the noise related to the environment of the signal capture and is often used for speech signals. However, it is impossible to make transparent the processing vis-à-vis the noise related to the environment of the sound recording, which poses a problem for music signal coding, in particular.
- the present invention improves the situation.
- the method according to the invention proposes a limitation of a distortion introduced by the post-processing to deliver a corrected output signal, by assigning to the corrected output signal: a current amplitude having an intermediate value between an amplitude value current signal of the post-processed signal and a corresponding current amplitude value of the decoded signal, or the current amplitude of the post-processed signal, according to the values respectively taken by the current amplitude of the post-processed signal and by the current amplitude corresponding decoded signal.
- a delay line is provided to ensure a temporal correspondence between the current amplitude of the post-processed signal and the corresponding current amplitude of the decoded signal.
- the method comprises the steps: defining a range of allowed amplitudes, the interval comprising a lower bound and an upper bound which are functions of a current amplitude value of the decoded (but not post-processed) signal, and for a corresponding current amplitude of the post-processed signal, assignment of a current amplitude value to the output signal, equal to the value of:
- the present invention proposes not to deviate from the decoded signal, beyond a certain tolerance, during the post-processing of the decoded signal.
- This embodiment can advantageously be implemented in the case where the received signal has been encoded by a scalar quantization coding, the decoder delivering quantized amplitude values which vary with each other in a discrete manner, the successive differences between the quantized values defining successive steps of quantification.
- the upper bound can be given by the addition of substantially half of the quantization step to the quantized value assigned to the current amplitude of the decoded signal
- the lower bound can be given by subtracting substantially half the quantization step from the quantized value assigned to the current amplitude of the decoded signal.
- scalar quantization coding is the so-called "pulse modulation coding" coding, delivering a coded index.
- pulse modulation coding coding
- a correspondence table can be provided, giving, for a current received index, a corresponding quantized value and a corresponding quantization step half, from which the current values of the lower and upper bounds can then be determined.
- FIG. 1 very schematically illustrates the general structure of a scalar quantization codec, whose decoder is followed by a post-processing and a module, within the meaning of the invention, of distortion limitation introduced by the postprocessing
- - Figure 2 schematically illustrates the structure of the distortion limitation module of Figure 1 and its 3 schematically illustrates distortion limitation steps within the meaning of the invention
- FIG. 4 very schematically illustrates the hardware structure of a distortion limitation module within the meaning of the invention. .
- the present invention is advantageously involved in the context of coding / decoding of the scalar quantization type.
- PCM type coding for "Pulse and Coding Modulation” - also called PCM in English, for "Pulse Code Modulation”
- each input sample is coded individually, without prediction.
- PCM type coding for "Pulse and Coding Modulation” - also called PCM in English, for "Pulse Code Modulation”
- This type of coding compresses signals sampled at 8 kHz, typically defined in a minimum frequency band of 300 to 3400 Hz, by a logarithmic curve which allows to obtain an almost constant signal-to-noise ratio for wide signal dynamics.
- the quantization step is approximately proportional to the amplitude of the signals.
- the initial signal S is first coded (module 10) in an encoder 13 and the resulting index sequence I MIC is represented on 8 bits per sample (see reference 15 of FIG. 1), which corresponds to 256 levels
- an original sample of the signal S to be encoded has an amplitude equal to -75. Therefore, this amplitude is in the range [-80, -65] of line 123 (or "level" 123) of the table.
- the coding of this information consists of delivering a final coded index, referenced I ' Mw in Figure 1 and in Table 1, which is equal to 0x51.
- VQ 32256 to all samples whose initial amplitude was in the interval [31744, 32767], that is, a total of 1024 possible values, which corresponds to a quantization step of 1024.
- the PCM compression is performed by a linear amplitude compression by segments.
- the 8 bits characterizing 256 quantized values are thus distributed as follows:
- the quantization step is multiplied by two (16, 32, 64, ...) in the transition from one segment to the next, and this, from the second segment.
- This encoding law therefore makes it possible to have a quantization precision of 12 bits (with a quantization step of 16) on the first two index segments 0 and 1 (Table 2). Then, the accuracy decreases by 1 bit each increment of the segment index (the quantization step being multiplied by two at each increment), as shown in Table 2 below.
- Table 2 is interpreted as follows. For example, if the amplitude of an original sample is -30000: the index of the associated segment "7" is coded on 3 bits, the sign "-" is coded on 1 bit (at 0) , and the remaining 4 bits (13, 12, 11 and 10) define the amplitude level in the index segment 7. Similarly, if the amplitude of an original sample is +4000:
- Table 3 below is the equivalent of Table 2, but for the G.711 standard as practiced in particular in the United States of America or in Japan (called " ⁇ law"), with in particular the no quantization and the maximum possible deviations E MAX between the quantized value VQ and the real value of the amplitude of the original sample.
- the final index the M ⁇ c received at the decoder makes it possible to determine, on the one hand, the quantized value VQ and, on the other hand, the segment index ID-SEG from which the quantization step can be deduced and hence the maximum amplitude of the coding error E MAX - It will also be noted that the index of the ID-GIS segment can be found also as a function of the position of the weight bit the strongest of the signal amplitude in the case of G.711 coding according to law A (Table 2).
- a specificity of the MIC coding is that the original sample and the decoded sample always have their amplitude in the same quantization interval: for the original sample, at any position in the meantime, and for the decoded sample, systematically in the middle of the interval.
- the decoded signal S ' MIC then undergoes post-processing filtering 16 (for example a denoising or the application of a perceptual post-filter).
- the resulting signal S POST is then processed by a module 20 within the meaning of the invention.
- the post-processing 16 (even if it is generally linear phase to preserve the waveform) may be too aggressive and alter the natural appearance of a speech signal.
- the decoder there is information on the original signal that can be used, within the meaning of the present invention, to limit the difference between the decoded and post-filtered signal S POST , on the one hand, and the original signal S, on the other hand.
- Module 20 ( Figure 1) allows, in the sense of the invention, to limit the distortion generated by the post-processing implemented decoding.
- a possible exemplary embodiment, described in detail below, is to impose that the distortion introduced by the post-processing 16 with respect to the decoded signal S ' MIC can not be greater than the maximum amplitude of the coding error.
- E M A X - It is then ensured that the post-filtered signal remains in the same quantization interval as the original signal.
- the overall distortion due to coding / decoding and post processing processing is limited, and in particular very close to the maximum coding distortion E M A X - This measurement also ensures that the energy distribution between successive samples and the overall waveform be well preserved.
- FIG. 2 An exemplary implementation of the invention is illustrated in FIG. 2.
- the module 21 calculates the decoded sample S' MW V by inverse quantization of the index I MW received.
- the module 22 performs the aforementioned post-processing. It will also be remembered that this operation generally introduces a delay.
- provision is made for a treatment in the sense of the invention which advantageously starts with a delay line (module 23) to which the received index I ' MW is also applied.
- the delay is adjusted so that the delayed index I ' M W_ DEL is time-aligned with the current sample delivered by the output S P OS T of post-processing 22.
- the module 25 determines the quantized value QV and the corresponding maximum coding error E MAX , for example from a table 24 which can comprise data from Table 1 above. .
- the data in Table 1 which can be used for the determination of the QV and E MAX parameters operated by the module 25 are shown in Table 4 below.
- Table 4 changes according to the quantized value QV to show that Table 4 is taken from Table 1 given above.
- Table 5 contains the same data as Table 4, but is ranked according to the index values I'MW_DEL- Table 5 then presents the respective parameters QV and E MAX according to a given index I ' MW _ DEL and can therefore constitute, for the standard G.711-law A, the content of the table 24 of FIG.
- Table 6 Table 7 below shows the data that can comprise a table 24 in a processing of the type shown in FIG. 2, in the context of coding according to the law ⁇ .
- the table 24 (can therefore include the data of Tables 5 or 7) can be stored hard in a memory of a module 20 ( Figure 1) within the meaning of the invention.
- the parameters E MAX and QV are calculated directly from the received index, without using a table 24, as follows.
- the module 25 can calculate the maximum coding error E MAX linked to this identifier segment ID-SEG, from a function of simple correspondence between the ID-SEG identifier and the E MAX parameter, this function being able to be constructed from: the existing function linking the ID-SEG identifier to the quantization step and the existing function linking the quantization step to the maximum coding error E M AX, in accordance with Tables 2 and 3 given previously.
- the module 26 checks whether the difference between the post-processed sample S POST and the just decoded sample without post-processing S ' MW does not exceed the value of the parameter E MAX found, in which case the post-processing has induced distortions that should be limited.
- the value of the sample S POST is then brought back to a value closer to the quantified value QV, so that the difference between the values S POST and QV remains below an authorized threshold.
- the module 26 operates, as follows, on the basis of: - a post-processed current sample S POST , the quantized value QV of the corresponding sample just decoded without postprocessing, and the maximum error of E MAX encoding found with this QV quantized value.
- FIG 3 details the operations of module 26 of Figure 2 in the form of a flowchart.
- the inputs of this module are thus the S POST post-processed samples, the corresponding quantized QV values and the corresponding maximum E MAX coding errors (step 31).
- steps 32 and 33 the limits, respectively lower LimiNF and upper Limsup of the quantization interval around the current quantized value QV, are determined.
- step 34 it is verified whether the post-processed sample S POST has a lower amplitude than the lower limit Limi NF .
- the temporary variable Tmp is set: either to the amplitude value of the sample S POST , or to that of the authorized lower limit LimiNF (if the amplitude SPOST is less than the limit Limi NF ).
- step 35 the output S OUT gives: either the unchanged value of the amplitude of the sample S POST (if it was already in the range delimited by the limits LimiNF and Limsup), - either the lower limit LimiNF (if the amplitude of the sample S POST was lower than the latter Limi N p), - or the upper limit Limsup (if the amplitude of the sample SPOST was superior to this last Limsup).
- the output signal S OUT always remains in the same quantization interval as the original signal S.
- the output signal is strictly brought back into the quantization interval of the original signal, delimited by: [S 'Mw ⁇ ⁇ MAX, S' Mw ⁇ ⁇ MAX ⁇ ⁇ ] -
- the distortion of the post-processing is limited compared to the decoded signal, and not necessarily compared to the original signal, depending on the type of coding / decoding used.
- an optional prior step 38 (illustrated in dotted lines for this purpose) can be provided to prevent the post-processing distortion limitation from being systematically applied. In certain cases, it is indeed advantageous to inhibit the treatment of FIG.
- the signal-to-noise ratio (noted RSB hereafter), obtained by the PCM coding / decoding, is substantially constant (at a level of about 38 dB) for wide signal dynamics.
- the RSB ratio is low and can even be negative at the beginning of the segment of the amplitude compression law.
- the output of the PCM decoder is then very "noisy" for low amplitude signals (for example in the case of silence between two sentences of a speech signal).
- the steps 32 to 35 do not are not implemented and the amplitude of the output samples S OUT directly takes the value of the amplitude of the post-filtered samples S POST (step 37).
- the value of the threshold S e is equal to 24 (in the scale, of course, tables given above).
- the treatment aimed at limiting the distortion is applied.
- the method according to the invention is finally implemented only for decoded and post-processed S POST signals whose amplitude is greater than the predetermined threshold value S e .
- the present invention is not limited to the embodiment described above by way of example; it extends to other variants.
- the distortion limitation module 20 is shown in FIG. 1 downstream of the post-processing module 16.
- it can be integrated directly into the post-processing module 16.
- this variant can be advantageous, particularly in the context of using recursive filters with infinite impulse response (or HR for Infinity Impulse Response). in English).
- HR filter the output sample of the filter depends on the previous outputs of this filter.
- the output of the HR filtering can directly take into account the values that were immediately modified by the module within the meaning of the invention.
- intervals were defined around the decoded value S '(which may be the quantized value QV in the case of scalar quantization coding / decoding of the type described herein. -before).
- this embodiment was described by way of non-limiting example. It can be provided, alternatively, to assign to the amplitude of the output signal S OUT the average (or more generally a weighted average) between the decoded value S 'and the post-processed amplitude value S POST , while allowing the direct assignment of the post-processed amplitude value SPOST if, for example, the latter SPOST is still in a chosen range.
- the present invention applies to any type of coding / decoding, beyond a coding according to the G.711 standard, and for example the embodiment described in detail above can be applied in particular to case of scalar quantization coding / decoding with any number of levels, followed by decoding of a linear phase postprocessing.
- the present invention also aims at a processing module 20 of a digital signal, this signal being decoded by an upstream decoder 14 (FIG. 1) and undergoing a noise reduction aftertreatment 16.
- This processing module 20 in the sense of the invention then comprises means 23, 24, 25, 26 ( Figure 2) for implementing the method of limiting a distortion introduced by the post-processing.
- this module 20 in the sense of the invention typically comprises, with reference to FIG.
- a ⁇ P processor cooperating with a memory block BM including a storage and / or working memory, as well as the aforementioned memory MEM as means for carrying out, in an exemplary embodiment, the delay line 23 and providing the delayed index I ' MW _ DEL -
- the memory block BM may further comprise a storage means (preferentially in read-only memory) of the correspondence table 24 of Figure 2, or a computer program for directly calculating the decoded value and the corresponding interval from the delayed index I ' MW _ DEL , according to the embodiment chosen.
- the module 20 may be independent or integrated in a noise reduction post-processing module.
- a storage memory of such a module 20 can advantageously also include a computer program comprising instructions for implementing the method in the sense of the invention, when these instructions are executed by a ⁇ P processor of the module.
- FIG. 3 may illustrate a flowchart representing the algorithm of such a computer program.
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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)
- Analogue/Digital Conversion (AREA)
Abstract
Description
Claims
Applications Claiming Priority (2)
Application Number | Priority Date | Filing Date | Title |
---|---|---|---|
FR0704901 | 2007-07-06 | ||
PCT/FR2008/051246 WO2009010672A2 (fr) | 2007-07-06 | 2008-07-04 | Limitation de distorsion introduite par un post-traitement au decodage d'un signal numerique |
Publications (2)
Publication Number | Publication Date |
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EP2162883A2 true EP2162883A2 (fr) | 2010-03-17 |
EP2162883B1 EP2162883B1 (fr) | 2012-09-05 |
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EP08806164A Active EP2162883B1 (fr) | 2007-07-06 | 2008-07-04 | Limitation de distorsion introduite par un post-traitement au decodage d'un signal numerique |
Country Status (6)
Country | Link |
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US (1) | US8571856B2 (fr) |
EP (1) | EP2162883B1 (fr) |
JP (1) | JP5179578B2 (fr) |
KR (1) | KR101470940B1 (fr) |
CN (1) | CN101816041B (fr) |
WO (1) | WO2009010672A2 (fr) |
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HUE039862T2 (hu) * | 2010-07-02 | 2019-02-28 | Dolby Int Ab | Audio dekódolás szelektív utószûréssel |
Family Cites Families (15)
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US5042069A (en) * | 1989-04-18 | 1991-08-20 | Pacific Communications Sciences, Inc. | Methods and apparatus for reconstructing non-quantized adaptively transformed voice signals |
JPH03116197A (ja) * | 1989-09-29 | 1991-05-17 | Matsushita Electric Ind Co Ltd | 音声復号化装置 |
JP3071800B2 (ja) * | 1990-02-23 | 2000-07-31 | 株式会社東芝 | 適応ポストフィルタ |
US5491514A (en) * | 1993-01-28 | 1996-02-13 | Matsushita Electric Industrial Co., Ltd. | Coding apparatus, decoding apparatus, coding-decoding apparatus for video signals, and optical disks conforming thereto |
JP4249414B2 (ja) * | 1999-10-08 | 2009-04-02 | 株式会社ケンウッド | ディジタル信号を補間する方法および装置 |
JP4689269B2 (ja) * | 2002-07-01 | 2011-05-25 | コーニンクレッカ フィリップス エレクトロニクス エヌ ヴィ | 静的スペクトルパワー依存型音響強調システム |
CN1666571A (zh) * | 2002-07-08 | 2005-09-07 | 皇家飞利浦电子股份有限公司 | 音频处理 |
US6839010B1 (en) * | 2002-12-27 | 2005-01-04 | Zilog, Inc. | Sigma-delta analog-to-digital converter with reduced quantization noise |
JP4311034B2 (ja) * | 2003-02-14 | 2009-08-12 | 沖電気工業株式会社 | 帯域復元装置及び電話機 |
ATE356405T1 (de) * | 2003-07-07 | 2007-03-15 | Koninkl Philips Electronics Nv | System und verfahren zur signalverarbeitung |
AU2003274864A1 (en) * | 2003-10-24 | 2005-05-11 | Nokia Corpration | Noise-dependent postfiltering |
US6950048B1 (en) * | 2004-04-02 | 2005-09-27 | Tektronix, Inc. | Dither system for a quantizing device |
US7787563B2 (en) * | 2004-12-08 | 2010-08-31 | Texas Instruments Incorporated | Transmitter for wireless applications incorporation spectral emission shaping sigma delta modulator |
US8180068B2 (en) * | 2005-03-07 | 2012-05-15 | Toa Corporation | Noise eliminating apparatus |
US20060217970A1 (en) * | 2005-03-28 | 2006-09-28 | Tellabs Operations, Inc. | Method and apparatus for noise reduction |
-
2008
- 2008-07-04 CN CN2008801061787A patent/CN101816041B/zh active Active
- 2008-07-04 WO PCT/FR2008/051246 patent/WO2009010672A2/fr active Application Filing
- 2008-07-04 KR KR1020107000183A patent/KR101470940B1/ko active IP Right Grant
- 2008-07-04 JP JP2010514083A patent/JP5179578B2/ja active Active
- 2008-07-04 US US12/667,908 patent/US8571856B2/en active Active
- 2008-07-04 EP EP08806164A patent/EP2162883B1/fr active Active
Non-Patent Citations (1)
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See references of WO2009010672A3 * |
Also Published As
Publication number | Publication date |
---|---|
CN101816041A (zh) | 2010-08-25 |
KR20100042251A (ko) | 2010-04-23 |
US8571856B2 (en) | 2013-10-29 |
CN101816041B (zh) | 2012-12-26 |
EP2162883B1 (fr) | 2012-09-05 |
US20100241427A1 (en) | 2010-09-23 |
KR101470940B1 (ko) | 2014-12-09 |
WO2009010672A2 (fr) | 2009-01-22 |
JP5179578B2 (ja) | 2013-04-10 |
JP2010532875A (ja) | 2010-10-14 |
WO2009010672A3 (fr) | 2009-03-05 |
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