EP0570362A4 - Digital speech decoder having a postfilter with reduced spectral distortion. - Google Patents
Digital speech decoder having a postfilter with reduced spectral distortion.Info
- Publication number
- EP0570362A4 EP0570362A4 EP19900913916 EP90913916A EP0570362A4 EP 0570362 A4 EP0570362 A4 EP 0570362A4 EP 19900913916 EP19900913916 EP 19900913916 EP 90913916 A EP90913916 A EP 90913916A EP 0570362 A4 EP0570362 A4 EP 0570362A4
- Authority
- EP
- European Patent Office
- Prior art keywords
- component
- coefficients
- postfilter
- providing
- parameters
- 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
- 230000003595 spectral effect Effects 0.000 title claims abstract description 27
- 238000000034 method Methods 0.000 claims abstract description 25
- 230000005284 excitation Effects 0.000 claims description 13
- 230000015572 biosynthetic process Effects 0.000 claims description 12
- 238000003786 synthesis reaction Methods 0.000 claims description 12
- 230000001131 transforming effect Effects 0.000 claims 5
- 238000001914 filtration Methods 0.000 claims 4
- 238000004519 manufacturing process Methods 0.000 claims 2
- 230000003044 adaptive effect Effects 0.000 abstract description 7
- 238000009499 grossing Methods 0.000 abstract description 6
- 230000002411 adverse Effects 0.000 abstract 1
- 239000013598 vector Substances 0.000 description 4
- 238000006243 chemical reaction Methods 0.000 description 1
- 230000001627 detrimental effect Effects 0.000 description 1
- 230000000694 effects Effects 0.000 description 1
- 238000001228 spectrum Methods 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
Definitions
- This invention relates generally to speech coders, and more particularly to digital speech coders that use postfilters to enhance the speech quality.
- Speech coders and decoders are known in the art. Some speech coders convert analog voice samples into digitized representations, and subsequently represent the spectral speech information through use of linear predictive coding. Other speech coders improve upon ordinary linear predictive coding (LPC) techniques by providing an excitation signal that is related to the original voice signal.
- LPC linear predictive coding
- U.S. Patent No. 4,817,157 describes a digital speech coder and decoder having an improved vector excitation source wherein a codebook of codebook excitation vectors is accessed to select a codebook excitation signal that best fits the available information, and is used to provide a synthesized speech signal from an LPC filter that closely represents the original.
- An adaptive spectral postfilter (which is typically intended to enhance the perceptual quality of the synthetic speech), and another is a post emphasis filter (which contributes brightness to the synthetic speech result).
- An adaptive spectral postfilter is typically of the general form:
- the denominator term in the above postfilter representation emphasizes the formants in the synthetic signal spectrum, while attenuating the spectral valleys. (In the two extremes, setting v - 0 results in an all-pass filter, while setting v - 1 results in a denominator term that is the same as the associated LPC filter.)
- the numerator term attempts to cancel the general spectral shape introduced by the denominator. In prior art applications, v is often set to about 0.8, and ⁇ to about 0.5.
- the numerator polynomial is only partially successful in tracking the spectral shape of the denominator (in effect, the spectral characteristic of the filter tilts with time), and that discrepancy typically manifests itself as a time varying modulation of the postfiltered speech brightness. Accordingly, a need exists for a method of postfiltering synthesized speech that will both enhance the perceptual quality of the synthetic speech, while simultaneously minimizing detrimental impact on speech brightness. Preferably, speech brightness itself will be better controlled as well.
- a postfilter which postfilter is characterized by a first and second component.
- the first component includes a set of coefficients. These coefficients are transformed into an alternate domain set of parameters, and then operated on to provide a modified set of parameters. These are then used to provide a set of coefficients that characterize the second component.
- Z transform (filter) coefficients that represent the first component are converted to the autocorrelation domain.
- a spectral smoothing techr ue that makes use of a bandwidth expansion function is then applied to the autocorrelation sequence, and the second component polynomial coefficients are calculated from the modified autocorrelation sequence via the Levinson recursion.
- the first component is then used as the denominator, and the second component as the numerator, in the above noted filter characteristic.
- the numerator polynomial is replaced by a spectrally smoothed version of the A(z v) polynomial.
- Format bandwidth expansion does not change the smoothed spectral envelope.
- the spectrally smoothed bandwidth expanded version of the A(z/v) polynomial effectively minimizes time varying spectral tilt and allows the numerator to adaptively track the general spectral shape of the denominator and cancel it out.
- an additional post emphasis filter can be used to afford more control over postfiltered speech brightness.
- This filter is a first order filter of the form
- H(z) 1 - uz "1 , where typically 0.2 ⁇ u ⁇ 0.5.
- Fig. 1 comprises a block diagrammatic depiction of a radio configured in accordance with the invention.
- a radio (100) embodying the invention includes an antenna (102) for receiving a speech coded radio frequency (RF) signal (101).
- An RF unit (103) processes the received signal to recover the speech coded information.
- This information is provided to a parameter decoder (105) that develops control parameters for various subsequent processes.
- An excitation source (104) as described above utilizes the parameters provided to it to create an excitation signal.
- This resultant excitation signal from the excitation source (104) is provided to an LPC filter (106) that yields a synthesized speech signal in accordance with the coded information.
- the synthesized speech signal is then pitch postfiltered (107) and spectrally postfiltered (108) to enhance the quality of the reconstructed speech.
- a post emphasis filter (109) can also be included to further enhance the resultant speech signal. (Additional details regarding the spectral postfilter (108) and the post emphasis filter (109) will be provided below.)
- the speech signal is then processed in an audio processing unit (111) and rendered audible by an audio transducer (112).
- the excitation source (104), LPC filter (106), pitch postfilter (107), adaptive spectral postfilter (108), and post emphasis filter (109) can all be provided through appropriate programming of a DSP (1 13).
- the adaptive spectral postfilter (108) is characterized by a first component (a denominator that is related to the filter characteristics of the LPC filter (106)) and a second component (a numerator that adaptively tracks the general spectral shape of the denominator to thereby cancel it out).
- a first component a denominator that is related to the filter characteristics of the LPC filter (106)
- a second component a numerator that adaptively tracks the general spectral shape of the denominator to thereby cancel it out.
- the general form of such a filter can be found described in an article entitled "Real-Time Vector APC Speech Coding at 4800 bps With Adaptive Postfiltering," by Chen and Gersho, which appeared in the April, 1987 edition of the Proceedings of The International Conference on Acoustics, Speech, and Signal Processing, at pages 2185-2188.
- the numerator is developed by applying spectral smoothing techniques to the denominator polynomial.
- Z transform coefficients that represent the denominator are converted to the autocorrelation domain.
- Examples of such conversions can be found in Markel, J.D. Gray, A.H., Jr.; Linear Prediction of Speech (Springer-Verlag, Berlin, Heidelberg, New York, 1976.)
- the spectral smoothing technique bandwidth expansion function is then applied to the autocorrelation sequence, with the numerator polynomial coefficients being calculated from the modified autocorrelation sequence via the Levinson recursion.
- the autocorrelation coefficients are multiplied by the following factors to provide the resultant numerator coefficients:
- the denominator and numerator are then used to characterize the adaptive spectral postfilter (108 ⁇ .
- the numerator polynomial is provided by a spectrally smoothed version of the denominator polynomial.
- the spectrally smoothed bandwidth expanded version of the denominator polynomial effectively minimizes time varying spectral tilt and allows the numerator to adaptively track the general spectral shape of the denominator and cancel it out.
- a bandwidth expansion factor (which specifies the degree of smoothing that is performed on the denominator) of about 1 ,200 Hz was used. ⁇
- the post emphasis filter (109) may be provided to afford more control over postfiltered speech brightness.
- This filter is a first order filter of the form
- H(z) 1 - uz "1 , where typically 0.2 ⁇ u ⁇ 0.5.
Landscapes
- 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)
- Filters That Use Time-Delay Elements (AREA)
- Transmission Systems Not Characterized By The Medium Used For Transmission (AREA)
Abstract
Description
Claims
Applications Claiming Priority (3)
Application Number | Priority Date | Filing Date | Title |
---|---|---|---|
US42292689A | 1989-10-17 | 1989-10-17 | |
US422926 | 1989-10-17 | ||
PCT/US1990/005190 WO1991006093A1 (en) | 1989-10-17 | 1990-09-17 | Digital speech decoder having a postfilter with reduced spectral distortion |
Publications (3)
Publication Number | Publication Date |
---|---|
EP0570362A4 true EP0570362A4 (en) | 1993-07-01 |
EP0570362A1 EP0570362A1 (en) | 1993-11-24 |
EP0570362B1 EP0570362B1 (en) | 1999-03-17 |
Family
ID=23676980
Family Applications (1)
Application Number | Title | Priority Date | Filing Date |
---|---|---|---|
EP90913916A Expired - Lifetime EP0570362B1 (en) | 1989-10-17 | 1990-09-17 | Digital speech decoder having a postfilter with reduced spectral distortion |
Country Status (8)
Country | Link |
---|---|
EP (1) | EP0570362B1 (en) |
JP (1) | JP3158434B2 (en) |
CN (1) | CN1078371C (en) |
AT (1) | ATE177867T1 (en) |
AU (1) | AU635342B2 (en) |
DE (1) | DE69033011T2 (en) |
ES (1) | ES2131498T3 (en) |
WO (1) | WO1991006093A1 (en) |
Families Citing this family (6)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
FR2729244B1 (en) * | 1995-01-06 | 1997-03-28 | Matra Communication | SYNTHESIS ANALYSIS SPEECH CODING METHOD |
FR2729246A1 (en) * | 1995-01-06 | 1996-07-12 | Matra Communication | SYNTHETIC ANALYSIS-SPEECH CODING METHOD |
FR2729247A1 (en) * | 1995-01-06 | 1996-07-12 | Matra Communication | SYNTHETIC ANALYSIS-SPEECH CODING METHOD |
JP2993396B2 (en) * | 1995-05-12 | 1999-12-20 | 三菱電機株式会社 | Voice processing filter and voice synthesizer |
DE19643900C1 (en) * | 1996-10-30 | 1998-02-12 | Ericsson Telefon Ab L M | Audio signal post filter, especially for speech signals |
US6137844A (en) * | 1998-02-02 | 2000-10-24 | Oki Telecom, Inc. | Digital filter for noise and error removal in transmitted analog signals |
Citations (1)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
EP0294020A2 (en) * | 1987-04-06 | 1988-12-07 | Voicecraft, Inc. | Vector adaptive coding method for speech and audio |
Family Cites Families (5)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
US4301329A (en) * | 1978-01-09 | 1981-11-17 | Nippon Electric Co., Ltd. | Speech analysis and synthesis apparatus |
US4617676A (en) * | 1984-09-04 | 1986-10-14 | At&T Bell Laboratories | Predictive communication system filtering arrangement |
JP2535833B2 (en) * | 1986-07-03 | 1996-09-18 | 日本電気株式会社 | Integrated circuit |
US4852169A (en) * | 1986-12-16 | 1989-07-25 | GTE Laboratories, Incorporation | Method for enhancing the quality of coded speech |
US4817157A (en) * | 1988-01-07 | 1989-03-28 | Motorola, Inc. | Digital speech coder having improved vector excitation source |
-
1990
- 1990-09-17 EP EP90913916A patent/EP0570362B1/en not_active Expired - Lifetime
- 1990-09-17 DE DE69033011T patent/DE69033011T2/en not_active Expired - Lifetime
- 1990-09-17 ES ES90913916T patent/ES2131498T3/en not_active Expired - Lifetime
- 1990-09-17 WO PCT/US1990/005190 patent/WO1991006093A1/en active IP Right Grant
- 1990-09-17 AT AT90913916T patent/ATE177867T1/en not_active IP Right Cessation
- 1990-09-17 JP JP51307390A patent/JP3158434B2/en not_active Expired - Lifetime
- 1990-09-17 AU AU64114/90A patent/AU635342B2/en not_active Expired
- 1990-10-15 CN CN90108435.2A patent/CN1078371C/en not_active Expired - Lifetime
Patent Citations (1)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
EP0294020A2 (en) * | 1987-04-06 | 1988-12-07 | Voicecraft, Inc. | Vector adaptive coding method for speech and audio |
Non-Patent Citations (1)
Title |
---|
ADVANCES IN SPEECH CODING-IEEE WORKSHOP ON SPEECH CODING FOR TELECOMMUNICATIONS 1991, DORDRECHT NL-VANCOUVER CANADA pages 69 - 79 GERSON, JASIUK 'Vector sum excited linear prediction' * |
Also Published As
Publication number | Publication date |
---|---|
CN1078371C (en) | 2002-01-23 |
EP0570362B1 (en) | 1999-03-17 |
AU6411490A (en) | 1991-05-16 |
JPH05500573A (en) | 1993-02-04 |
ATE177867T1 (en) | 1999-04-15 |
CN1051101A (en) | 1991-05-01 |
WO1991006093A1 (en) | 1991-05-02 |
AU635342B2 (en) | 1993-03-18 |
EP0570362A1 (en) | 1993-11-24 |
DE69033011T2 (en) | 2001-10-04 |
DE69033011D1 (en) | 1999-04-22 |
JP3158434B2 (en) | 2001-04-23 |
ES2131498T3 (en) | 1999-08-01 |
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