INJECTION HIGH FREQUENCY NOISE INTO PULSE EXCITATION FOR LOW BIT RATE CELP

Abstract: A speech-coding system provides improved speech coding by injecting high-frequency noise into an output of a pulse codebook. A filtered noise is generated by passing a high frequency noise signal through a high pass filter. The filtered high frequency noise is injected into the pulse output of the codebook through convolution. The combined noise signal and pulse output generates a perceptually improved encoded speech signal.

Description: INJECTING HIGH FREQUENCY NOISE INTO PULSE EXCITATION FOR LOW BIT RATE CELP BACKGROUND OF THE INVENTION 1. Cross Reference to Related Applications. This application claims the benefit of Provisional Application No 60/233,043 filed on September 15, 2000 The following co-pendmg and commonly assigned U S patent applications have been filed on the same day as this application All of these applications relate to and further describe other aspects of the embodiments disclosed in this application and are incorporated by reference m their entirety United States Patent Application Serial Number , "SELECTABLE MODE VOCODER SYSTEM," Attorney Reference Number 98RSS365CIP (10508 4), filed on September 15, 2000, and is now United States Patent Number United States Patent Application Serial Number , "SHORT TERM ENHANCEMENT IN CELP SPEECH CODING," Attorney Reference Number 00CXT0666N (10508 6), filed on September 15, 2000, and is now United States Patent Number United States Patent Application Serial Number , "SYSTEM OF DYNAMIC PULSE POSITION TRACKS FOR PULSE-LIKE EXCITATION IN SPEECH CODING," Attorney Reference Number 00CXT0573N (10508 7), filed on September 15, 2000, and is now United States Patent Number United States Patent Application Serial Number , "SPEECH CODING SYSTEM WITH TIME-DOMAIN NOISE ATTENUATION," Attorney Reference Number 00CXT0554N (10508 8), filed on September 15, 2000, and is now United States Patent Number United States Patent Application Serial Number , "SYSTEM FOR AN ADAPTIVE EXCITATION PATTERN FOR SPEECH CODING," Attorney Reference Number- 98RSS366 (10508 9), filed on September 15, 2000, and is now United ' States Patent Number . United States Patent Application Serial Number , "SYSTEM FOR ENCODING SPEECH INFORMATION USING AN ADAPTIVE CODEBOOK WITH DIFFERENT RESOLUTION LEVELS," Attorney Reference Number: 00CXT0670N (10508 13), filed on September 15, 2000, and is now United States Patent Number United States Patent Application Serial Number , "CODEBOOK TABLES FOR ENCODING AND DECODING," Attorney Reference Number 00CXT0669N (10508 14), filed on September 15, 2000, and is now United States Patent Number . United States Patent Application Serial Number , "BIT STREAM PROTOCOL FOR TRANSMISSION OF ENCODED VOICE SIGNALS," Attorney Reference Number 00CXT0668N (10508 15), filed on September 15, 2000, and is now United States Patent Number . United States Patent Application Serial Number , "SYSTEM FOR FILTERING SPECTRAL CONTENT OF A SIGNAL FOR SPEECH ENCODING," Attorney Reference Number. 00CXT0667N (10508 16), filed on September 15, 2000, and is now United States Patent Number United States Patent Application Seπal Number , "SYSTEM FOR ENCODING AND DECODING SPEECH SIGNALS," Attorney Reference Number: 00CXT0665N (10508.17), filed on September 15, 2000, and is now United States Patent Number United States Patent Application Serial Number , "SYSTEM FOR SPEECH ENCODING HAVING AN ADAPTIVE FRAME ARRANGEMENT," Attorney Reference Number 98RSS384CIP (10508 18), filed on September 15, 2000, and is now United States Patent Number United States Patent Application Serial Number , "SYSTEM FOR IMPROVED USE OF PITCH ENHANCEMENT WITH SUB CODEBOOKS," Attorney Reference Number 00CXT0569N (10508 19), filed on September 15, 2000, and is now United States Patent Number 2. Field of the Invention. This invention relates to speech coding, and more particularly, to a system that enhances the perceptual quality of digital processed speech 3. Related Art. Speech synthesis is a complex process that often requires the transformation of voiced and unvoiced sounds into digital signals To model sounds, the sounds are sampled and encoded into a discrete sequence The number of bits used to represent the sounds can determine the perceptual quality of synthesized sound or speech A pooi quality replica can drown out voices with noise, lose clarity, or fail to capture the inflections, tone, pitch, or co-articulations that can create adjacent sounds In one technique of speech synthesis known as Code Excited Linear Predictive Coding (CELP) a sound track is sampled into a discrete waveform before being digitally processed The discrete waveform is then analyzed according to certain select criteria Criteria such as the degree of noise content and the degree of voice content can be used to model speech through linear functions in real and in delayed time These linear functions can capture information and predict future waveforms The CELP coder structure can produce high quality reconstructed speech However, coder quality can drop quickly when its bit rate is reduced To maintain a high coder quality at a low bit rate, such as 4 Kbps, additional approaches must be explored This invention is directed to providing an efficient coding system of voiced speech and to a method that accurately encodes and decodes the perceptually important features of voiced speech SUMMARY This invention is a system that seamlessly improves the encoding and the decoding of perceptually important features of voiced speech The system uses modified pulse excitations to enhance the perceptual quality of voiced speech at high frequencies The system includes a pulse codebook, a noise source, and a filter The filter connects an output of the noise source to an output of the pulse codebook The noise source may generate a white noise, such as a Gaussian white noise, that is filtered by a high pass filter The pass band of the filter passes a selected portion of the white Gaussian noise The filtered noise is scaled, windowed, and added to a single pulse to generate an impulse response that is convoluted with the output of the pulse codebook In another aspect, an adaptive high-frequency noise is injected into the output of the pulse codebook The magnitude of the adaptive noise is based on a selectable criteria such as the degree of noise like content in a high-frequency portion of a speech signal, the degree of voice content in a sound track, the degree of unvoiced content in a sound track, the energy content of a sound track, the degree of periodicity in a sound track, etc The system generates different energy or noise levels that targets one or more of the selected criteria Preferably, the noise levels model one or more important perceptual features of a speech segment Other systems, methods, features and advantages of the invention will be or will become apparent to one with skill in the art upon examination of the following figures and detailed description It is intended that all such additional systems, methods, features and advantages be included within this description, be within the scope of the invention, and be protected by the accompanying claims BRI EF DESCRIPTION OF THE FIGURES The components in the figures are not necessarily to scale, emphasis instead being placed ,upon illustrating the principles of the invention Moreover, in the figures, like reference numerals designate corresponding parts throughout the different views FIG 1 is a partial block diagram of a speech communication system that may be incorporated in an extended Code Excited Linear Prediction System (eX-CELPS) FIG 2 illustrates a fixed codebook of FIG 1 FIG 3 illustrates sectional views of a part of a pulse of the fixed codebook of FIG 1 in the time-domain FIG 4 illustrates the impulse response of a first pulse Pi of FIG 3 in the frequency-domain FIG 5 illustrates the injection of a modified high frequency noise into the pulse excitations of FIG 3 in the time-domain FIG 6 is a flow diagram of an enhancement of FIG 1 FIG 7 illustrates a discrete implementation of the enhancement of FIG 1 The dashed lines drawn in FIGS 1 , 2, and 6 represent direct and indirect connections As shown in FIG 2, the fixed codebook 102 can include one or more subcodebooks Similarly, the dashed lines of FIG 6 illustrate that other functions can occur before or after each illustrated step DETAILED DESCRIPTION Pulse excitations typically can produce better speech quality than conventional noise excitation, for voiced speech Pulse excitations track the quasi-periodic time- domain signal of voiced speech at low frequencies At high frequencies, however, low bit rate pulse excitations often cannot track the perceptual "noisy effect" that accompanies voiced speech This can be a problem especially at very low bit rates such as 4 Kbps or lower rates for example where pulse excitations must track, not only the periodicity of voiced speech, but also the accompanying "noisy effects" that occur at higher frequencies FIG 1 is a partial block diagram of a speech communication system 100 that may be incorporated in a variant of a Code Excited Linear Prediction System (CELPS) known as the extended Code Excited Linear Prediction System (eX- CELPS) Conceptually, eX-CELP achieves toll quality at a low bit rate by emphasizing the perceptually important features of a sampled input signal (i e , a voiced speech signal) while de-emphasizing the auditory features that are not perceived by a listener Using a process of linear predictions, this embodiment can represent any sample of speech The short-term prediction of speech s at an instant n can be approximated by Equation 1 s(ri) * a / s n - 1) + a 2 s(n - 2) + + a p s(n - p) (Equation 1) where a / a 2 a p are Linear Prediction Coding (LPC) coefficients and p is the Linear Prediction Coding order The difference between the speech sample and the predicted speech sample is known as the prediction residual r(n) having a similar periodicity as speech signal s(n) The prediction residual r(n) can be expressed as r(n) = s(n) - ciι s(n - 1) - a 2 s(n - 2) - - a p s(n - p) (Equation 2) which can be re-wπtten as s(n) = r(n) + a / s(n - l) + a 2 s(n - 2) + + a p s(n - p) (Equation 3) A closer examination of Equation 3 reveals that a current speech sample can be broken down into a predictive portion a ι s(n - \) + a 2 s(n - 2) + + a p s(n - p) and an innovative portion r(ή) In some cases, the coded innovation portion is called the excitation signal or e(n) 106 It is the filtering of the excitation signal e(n) 106 by a synthesizer or a synthesis filter 108 that produces the reconstructed speech signal s'(n) 1 10 To ensure that voiced and unvoiced speech segments are accurately reproduced, the excitation signal e(n) 106 is created through a linear combination of the outputs from an adaptive codebook 1 12 and a fixed codebook 102 The adaptive codebook 1 12 generates signals that represent the periodicity of the speech signal s(n) In this embodiment, the contents of the adaptive codebook 1 12 are formed from previously reconstructed excitations signals e(ή) 106. These signals repeat the content of a selectable range of previously sampled signals that lie within adjacent subframes. The content is stored in memory. Due to the high-degree of correlation that exists between the current and previous adjacent subframes, the adaptive codebook 1 12 tracks signals through selected adjacent subframes and then uses these previously sampled signals to generate the entire or a portion of the current excitation signal e(n) 106. The second codebook used to generate the entire or a portion of the excitation signal e(n) 106 is the fixed codebook 102. The fixed codebook primarily contributes the non-predictable or non-periodic portion of the excitation signal e(n) 106. This contribution improves the approximation of the speech signal s(n) when the adaptive codebook 112 cannot effectively model non-periodic signals. When noise-like structures or non-periodic signals exist in a sound track because of rapid frequency variations in voiced speech or because transitory noise-like signals mask voiced speech, for example, the fixed codebook 102 produces a best approximation of these non-periodic signals that cannot be captured by the adaptive codebook 1 12. The overall objective of the selection of codebook entries in this embodiment is to create the best excitations that approximate the perceptually important features of a current speech segment. To improve performance,' a modular codebook structure is used in this embodiment that structures the codebooks into multiple sub codebooks. Preferably, the fixed codebook 102 is comprised of at least three sub codebooks 202 - 206 as illustrated in FIG. 2. Two of the fixed sub codebooks are pulse codebooks 202 and 204 such as a 2-pulse sub codebook and a 3-pulse sub codebook. The third codebook 206 may be a Gaussian codebook or a higher-pulse sub codebook. Preferably, the level of coding further refines the codebooks, particularly defining the number of entries for a given sub code book. For example, in this embodiment, the speech coding system differentiates "periodic" and "non-periodic" frames and employs full-rate, half-rate, and eighth-rate coding. Table 1 illustrates one of the many fixed sub codebook sizes that may be used for "non-periodic fames," where typical parameters, such as pitch correlation and pitch lag, for example, can change rapidly. Table 1 : Fixed Codebook Bit Allocation for Non-periodic Frames In "periodic frames," where a highly periodic signal is perceptually well represented with a smooth pitch track, the type and size of the fixed sub codebooks may vary from the fixed codebooks used in the "non-periodic frames." Table 2 illustrates one of the many fixed sub codebook sizes that may be used for "periodic fames." Table 2: Fixed Codebook Bit Allocation for Periodic Frames Other details of the fixed codebooks that may be used in a Selective Mode Vocoder (SMV) are further explained in the co-pending patent application entitled: "System of Encoding and Decoding Speech Signals" by Yang Gao, Adil Beyassine, Jes Thyssen, Eyal Shlomot, and Huan-yu Su that was previously incorporated by reference. Following a search of the fixed sub codebooks that yields the best output signals, some enhancements h>, h 2 , h}, . . . h n are convoluted with the outputs of the pulse sub codebooks to enhance the perceptual quality of the modeled signal. These enhancements preferably track select aspects of the speech .segment and are calculated from subframe to subframe. A first enhancement hi is introduced by injecting a high frequency noise into the pulse outputs that are generated from the pulse sub codebooks It should be noted that the high frequency enhancement hi generally is performed only on pulse sub codebooks and not on the Gaussian sub codebooks FIG 3 illustrates an exemplary output Y p (n) of a fixed pulse sub codebook To simplify the explanation, only three output pulses P / , P , and Pj 302 - 306 are illustrated in a single subframe Of course, any number of pulses P„ can be enhanced in a single or multiple subframes The three pulses Pi, P , and Rj 302 - 306 are positioned within a sub frame which has an exemplary time interval between 5 - 10 milliseconds In the frequency-domain, pulses Pi, P , and E 3 302 - 306 have a flat magnitude and a substantially linear phase (the magnitude and phase of Pi in the frequency-domain are illustrated in FIG 4) In the h t enhancement, a time-domain high frequency noise signal is added to Pi, P , and Rj 302 - 306 by convoluting Pi, P , and P} with an hι(ή) The product of the convolution is shown in FIG 5 FIG 6 is a flow diagram of the hi enhancement that can be convoluted with the excitation output of any pulse codebook to enhance the perceptual quality of a reconstructed speech signal s'(n) At step 602, a noise source generates a white Gaussian noise X(n) Preferably, the white Gaussian noise has a substantially flat magnitude in the frequency-domain At step 604, the white Gaussian noise X(n) may be filtered by a high-pass filter The cut-off frequency of the high pass filter may be defined by the desired perceptual qualities of the speech segment s( ) At step 606, the filtered noise X^(n) is scaled by a programmable gam factor g n that also can be a fixed or an adaptive gain factor in alternative embodiments At step 608, the noise X {") • gn 1S windowed with a smooth window W(n) (e g , a half Hamming window) of length L of samples vv(.) Preferably, the window W(n) attenuates the noise X^(n) • g n to a length oϊ ι(n) At steps 610 and 612, the modified noise is injected into the output Y p (n) of the pulse sub codebook as illustrated in FIG 5 and Equations 4 and 5 Preferably, delta of n of Equation 4, δ(n), is a single unit pulse that has a value of one at n = 0 and has a value of zero at all other values of n (I e , n ≠ 0) hι(n) = X (n) • g„ • W(n) + δ(n) (Equation 4) Y' p (n) = h,(n) * Y p (n) (Equation 5) Of course, the first enhancement h/ also can be implemented in the discrete-domain through a convolver having at least two ports or means 702 comprising a digital controller (1 e , a digital signal processor), one or more enhancement circuits, one or more digital filters, or other discrete circuitry, for example These implementations illustrated in FIG 7 can be written as follows Y' p (z) = H,(z) * Y p (z) (Equation 6) From the foregoing description it should be apparent that the addition of a decaying noise to an output of a pulse codebook also could be added prior to the occurrence of a pulse output Preferably, memory retains the hi enhancement of one or more previous subframes When hi is not generated before the occurrence of a pulse, a selected previous hi enhancement can be convoluted with the pulse codebook output before the occurrence of the pulse output The invention is not limited to a particular coding technology Any perceptual coding technology can be used including a Code Excited Linear Prediction System (CELP) and an Algebraic Code Excited Linear Prediction System (ACELP) Furthermore, the invention should not be limited to a closed-loop search used in an encoder The invention may also be used as a pulse processing method in a decoder Furthermore, prior to a search of the pulse sub codebooks, the hi enhancement may be incorporated within or made unitary with the sub codebooks or the synthesis filter 108 Many other alternatives are also possible For example, the noise energy can be fixed or adaptive In an adaptive noise embodiment, the invention can differentiate voiced speech using different criteria including the degree of noise like content in a high frequency portion of voiced speech, the degree of voice content in a sound track, the degree of unvoiced content in a sound track, the energy content in a sound track, the degree of periodicity in a sound track, etc , for example, and generate different energy or noise levels that target one or more selected criteria Preferably, the noise levels model one or more important perceptual features of a speech segment The invention seamlessly provides an efficient coding system and a method that improves the encoding and the decoding of perceptually important features of speech signals The seamless addition of high frequency noise to an excitation develops a high perceptual quality sound that a listener can come to expect in a high frequency range The invention may be adapted to post-processing technology and may be integrated within or made unitary with encoders, decoders, and codecs While various embodiments of the invention have been described, it will be apparent to those of ordinary skill in the art that many more embodiments and implementations are possible that are within the scope of this invention Accordingly, the invention is not to be restricted except in light of the attached claims and their equivalents

Claims: What is claimed is: 1. A speech communication system comprising: a first codebook that characterizes a speech excitation segment; a second codebook that characterizes a speech excitation segment; an convolver electrically connected to an output of the second codebook; and a synthesizer electrically connected to an output of the convolver and an output of the first codebook, the convolver being configured to inject a high frequency noise into an output of the second codebook. 2. A speech coding system comprising: a first codebook that characterizes a speech excitation segment; a second codebook that characterizes a speech excitation segment; a convolver connected to an output of the second codebook; and a synthesizer connected to an output of the convolver and an output of the first codebook, the convolver being configured to inject a high frequency noise into an output of the second codebook 3. The system of claim 2 where the first codebook comprises an adaptive codebook. 4, The system of claim 2 where the second codebook comprises a fixed codebook. 5. The system of claim 2 where the convolver comprises at least a two- port device configured to convolve two signals. 6. The system of claim 2 where the convolver comprises a high pass filter connected to a white noise source, the high pass filter being configured to pass a generated white noise. 7. The system of claim 2 where the convolver is configured to convolve an impulsive response containing a modified noise and an output signal produced by the second codebook. 8. The system of claim 2 where the synthesizer comprises a synthesis filter. 9. The system of claim 2 further comprising a scalar where the convolver is connected to the output of the second codebook and an input of the scalar. 10 The system of claim 2 where the system comprises a Code Excited Linear Prediction System. 1 1. The system of claim 2 where the system comprises an extended Code Excited Linear Prediction System. 12. The system of claim 2 where the convolver comprises a white noise source. 13. The system of claim 2 where the convolver injects the high frequency noise into an output of a pulse codebook. 14. The system of claim 2 where the convolver is configured to inject a modified white noise into the output of the second codebook. 15. The system of claim 14 where the convolver comprises an enhancement circuit configured to inject the modified white noise. 16. The system of claim 2 where the noise comprises an adaptive noise. 17. The system of claim 2 where the noise comprises a fixed noise. 18. The system of claim 2 where the first and the second codebooks, the convolver, and the synthesizer are provided in at least one of an encoder and a decoder. 19. A speech coding system comprising: a fixed codebook that characterizes a speech segment; an adaptive codebook that characterizes the speech segment; means configured to inject a high frequency noise into an output of the fixed codebook; and a synthesis filter connected to an output of the means. 20. The system of claim 19 where the means convolves a windowed high frequency noise, 21. The system of claim 19 where the means comprises a filter. 22. The system of claim 19 where the means comprises a high-pass filter. 23. The system of claim 19 where the means comprises a convolver. 24. The system of claim 19 where the means is connected to the output of the fixed codebook and an input of a summing circuit. 25. The system of claim 19 where the means and the fixed codebook are a unitary device. 26. The system of claim 19 where the means and the synthesis filter are a unitary device. 27. A method that improves speech coding comprising: forming an excitation signal by selecting an output from a pulse codebook; generating a decaying high frequency noise; and combining the high frequency noise with the output from the pulse codebook to produce an excitation that generates a speech segment. 28. The method of claim 27 where the pulse codebook comprises a fixed pulse codebook. 29. The method of claim 27 further comprising filtering the combined signals with a synthesis filter. 30. The method of claim 27 where the act of combining comprises convolving. 31. The method of claim 27 where the act of generating a decaying high frequency noise comprises generating a white noise, filtering the white noise with a high pass filter, and windowing a filtered noise with a smooth window. 32. The method of claim 31 where the window comprises a programmable window. 33. The method of claim 27 further comprising filtering the excitation with a synthesis filter.

Inventors: GAO YANG, GAO, YANG
Applicants: CONEXANT SYSTEMS INC, CONEXANT SYSTEMS, INC.
Application year: 2001
Main classification: G10L
IPC codes: G10L19, G10L21