`US007151802Bl
`
`(12) United States Patent
`Bessette et al.
`
`(IO) Patent No.:
`(45) Date of Patent:
`
`US 7,151,802 B1
`Dec. 19,2006
`
`(54)
`
`(75)
`
`IDGH FREQUENCY CONTENT
`RECOVERING METHOD AND DEVICE FOR
`OVER-SAMPLED SYNTHESIZED
`WIDEBM'D SIGNAL
`
`Inventors: Bruno Bessette, Rock Forest (CA);
`Redwau Salami, Sherbrooke (CA);
`Roeh Lefebvre, Canton de Magog
`(CA)
`
`(73) Assignee: Voiceage Corporation, Quebec (CA)
`( * ) Notice:
`
`Subject to any disclaimer, the term of this
`patent is extended or adjusted tu1der 35
`U.S.C. 154(b) by 0 days.
`
`(21) AppL No.:
`
`09/830,332
`
`(22) PCT Filed:
`
`Oct. 27, 1999
`
`(86) PCT No.:
`
`PCT/CA99/00990
`
`§ 371 (c)(l),
`(2), (4) Date:
`
`Jul. 23, 2001
`
`(87) PCT Pub. No.: W000/25305
`
`PCT Pub. Date: May 4, 2000
`
`(30)
`Foreign Application Priority Data
`Oct. 27, 1998
`(CA)
`.............. ...................... 2252170
`
`(51)
`
`Int. Cl.
`H04L 27100
`(2006.0])
`GJOL 19/02
`(2006.01)
`GJOL 11/04
`(2006.01)
`(52) U.S. Cl. .................. ..... 375/259; 704/203; 704/207
`(58) Field of Classi_fication Searcb ................ 375/259;
`455/65; 704/202, 203-207
`See application file for complete search history.
`
`(56)
`
`R eferences C ited
`
`U.S. PATENT DOCUMENTS
`5,113,262 A * 5/ 1992 Strolle et al. ................. 386/27
`5,235,669 A * 8/ 1993 Ordentlich et al. ......... 704/200
`(Continued)
`
`FOREIGN PATENT DOCUMENTS
`
`EP
`EP
`
`0545386 A2
`0838804 A2
`
`6/ 1993
`4/ 1998
`
`(Continued)
`
`OTHER PUBLICATIONS
`
`Holger Carl and Ulrich Heute, "Bandwidth Enhancement of Nar·
`row· Band Speech Signals," Signal Processing VU: Theories and
`Applications. vol. 11. pp. 1178·118 !.
`
`(Continued)
`
`Primary Examiner- Kl1ai Tran
`Assistant Examiner- Cicely Ware
`(74) Attorney, Agent, or Firm- Darby & Darby
`
`(57)
`
`ABSTRACT
`
`In a method and device tor recovering the high freq11ency
`content of a wideband signal previously down-sampled, and
`for injecting this high frequency content in an over-sampled
`synthesized version of the wideband signal to produce a
`fiU -spectn1m synthesized wideband signal, a random noise
`generator produces a noise sequence having a given spec(cid:173)
`trum. A spectral shaping unit spectrally shapes the noise
`sequence in relation to linear prediction filter coefficients
`related to the down·sam.pled wideband signal. A signal
`injection circuit finally jnjects the spectrally-shaped noise
`sequence in the over-sampled synthesized signal version to
`thereby produce the full-spectmm synthesized wideband
`signal,
`
`54 C laims, 4 Drawing Sheets
`
`200
`
`/
`
`
`
`OUTPUT
`SPE£CH
`
`ZTE EXHIDIT 1001
`
`Page 1 of 20
`
`
`
`US 7,151,802 B1
`Page 2
`
`U.S. PATENT DOCUMENTS
`5,394,473 A *
`5,444,816 A
`5,455,888 A *
`5,581,652 A *
`5,699,482 A
`5,701,392 A
`5,754,976 A
`5,956,624 A *
`5,978,759 A
`5,999,897 A *
`6,134,373 A *
`
`2/1995 Davidson ................. 704/200.1
`8/1995 Adoul eta!. ............... 395/2.28
`10/1995 Iyengar eta!. .............. 704/203
`12/1996 Abe et al .................... 704/222
`12/1997 Adoul eta!. ............... 395/2.28
`12/1997 Adoul eta!. ............... 395/2.28
`511998 Adoul et a!. ................ 704/223
`9/1999 Hunsinger eta!. ............ 455/65
`11/1999 Tsushima et a!.
`12/1999 Yeldener .................... 704/207
`10/2000 Strolle eta!. .................. 386/9
`
`FOREIGN PATENT DOCUMENTS
`
`JP
`JP
`
`08-123495
`08-248997
`
`5/1996
`9/1996
`
`OTHER PUBLICATIONS
`
`Yan Ming Cheng eta!., "Statistical Recovery ofWideband Speech
`from Narrowband Speech," IEEE Transactions on Speech and
`Audio Processing, vol. 2, No. 4, pp. 544-548.
`* cited by examiner
`
`Page 2 of 20
`
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`Page 5 of 20
`
`
`
`U.S. Patent
`
`Dec. 19,2006
`
`Sheet 4 of 4
`
`US 7,151,802 Bl
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`Page 6 of 20
`
`(cid:173)
`
`
`US 7,151,802 B1
`
`1
`HIGH FREQUENCY CONTENT
`RECOVERING METHOD AND DEVICE FOR
`OVER-SAMPLED SYNTHESIZED
`WIDEBAND SIGNAL
`
`BACKGROUND OF THE INVENTION
`
`1. Field of the Invention
`The present invention relates to a method and device for
`recovering a high frequency content of a wideband signal
`previously down-sampled, and for injecting this high fre(cid:173)
`quency content in an over-sampled synthesized version of
`the down-sampled wideband signal to produce a full-spec(cid:173)
`trum synthesized wideband signal.
`2. Brief Description of the Prior Art
`The demand for efficient digital wideband speech/audio
`encoding techniques with a good subjective quality/bit rate
`trade-off is increasing for numerous applications such as
`audio/video teleconferencing, multimedia, and wireless
`applications, as well as Internet and packet network appli- 20
`cations. Until recently, telephone bandwidths filtered in the
`range 200--3400 Hz were mainly used in speech coding
`applications. However, there is an increasing demand for
`wideband speech applications in order to increase the intel(cid:173)
`ligibility and naturalness of the speech signals. A bandwidth 25
`in the range 50-7000 Hz was found sufficient for delivering
`a face-to-face speech quality. For audio signals, this range
`gives an acceptable audio quality, but still lower than the CD
`quality which operates on the range 20-20000 Hz.
`A speech encoder converts a speech signal into a digital 30
`bitstream which is transmitted over a communication chan(cid:173)
`nel (or stored in a storage medium). The speech signal is
`digitized (sampled and quantized with usually 16-bits per
`sample) and the speech encoder has the role of representing
`these digital samples with a smaller number of bits while 35
`maintaining a good subjective speech quality. The speech
`decoder or synthesizer operates on the transmitted or stored
`bit stream and converts it back to a sound signal.
`One of the best prior art techniques capable of achieving
`a good quality/bit rate trade-off is the so-called Code Excited 40
`Linear Prediction (CELP) technique. According to this tech(cid:173)
`nique, the sampled speech signal is processed in successive
`blocks of L samples usually called frames where L is some
`predetermined number (corresponding to 10--30 ms of
`speech). In CELP, a linear prediction (LP) synthesis filter is 45
`computed and transmitted every frame. The L-sample frame
`is then divided into smaller blocks called subframes of size
`ofN samples, where L=kN and k is the number of subframes
`in a frame (N usually corresponds to 4-10 ms of speech). An
`excitation signal is determined in each subframe, which 50
`usually consists of two components: one from the past
`excitation (also called pitch contribution or adaptive code(cid:173)
`book) and the other from an innovative codebook (also
`called fixed codebook). This excitation signal is transmitted
`and used at the decoder as the input of the LP synthesis filter 55
`in order to obtain the synthesized speech.
`An innovative codebook in the CELP context, is an
`indexed set of N-sample-long sequences which will be
`referred to as N-dimensional codevectors. Each codebook
`sequence is indexed by an integer k ranging from 1 to M
`where M represents the size of the codebook often expressed
`as a number of bits b, where M=2 6
`.
`To synthesize speech according to the CELP technique,
`each block of N samples is synthesized by filtering an
`appropriate codevector from a codebook through time vary- 65
`ing filters modeling the spectral characteristics of the speech
`signal. At the encoder end, the synthesis output is computed
`
`10
`
`2
`for all, or a subset, of the codevectors from the codebook
`(codebook search). The retained codevector is the one
`producing the synthesis output closest to the original speech
`signal according to a perceptually weighted distortion mea-
`5 sure. This perceptual weighting is performed using a so(cid:173)
`called perceptual weighting filter, which is usually derived
`from the LP synthesis filter.
`The CELP model has been very successful in encoding
`telephone band sound signals, and several CELP-based
`standards exist in a wide range of applications, especially in
`digital cellular applications. In the telephone band, the sound
`signal is band-limited to 200-3400 Hz and sampled at 8000
`samples/sec. In wideband speech/audio applications, the
`15 sound signal is band-limited to 50-7000Hz and sampled at
`16000 samples/sec.
`Some difficulties arise when applying the telephone-band
`optimized CELP model to wideband signals, and additional
`features need to be added to the model in order to obtain high
`quality wideband signals. Wideband signals exhibit a much
`wider dynamic range compared to telephone-band signals,
`which results in precision problems when a fixed-point
`implementation of the algorithm is required (which is essen-
`tial in wireless applications). Further, the CELP model will
`often spend most of its encoding bits on the low-frequency
`region, which usually has higher energy contents, resulting
`in a low-pass output signal. To overcome this problem, the
`perceptual weighting filter has to be modified in order to suit
`wide band signals, and pre-emphasis techniques which boost
`the high frequency regions become important to reduce the
`dynamic range, yielding a simpler fixed-point implementa(cid:173)
`tion, and to ensure a better encoding of the higher frequency
`contents of the signal. Further, the pitch contents in the
`spectrum of voiced segments in wideband signals do not
`extend over the whole spectrum range, and the amount of
`voicing shows more variation compared to narrow-band
`signals. Thus, it is important to improve the closed-loop
`pitch analysis to better accommodate the variations in the
`voicing level.
`Some difficulties arise when applying the telephone-band
`optimized CELP model to wideband signals, and additional
`features need to be added to the model in order to obtain high
`quality wideband signals.
`As an example, in order to improve the coding efficiency
`and reduce the algorithmic complexity of the wideband
`encoding algorithm, the input wideband signal is down(cid:173)
`sampled from 16kHz to around 12.8 kHz. This reduces the
`number of samples in a frame, the processing time and the
`signal bandwidth below 7000Hz to thereby enable reduction
`in bit rate down to 12 kbit/s while keeping very high quality
`decoded sound signal. The complexity is also reduced due to
`the lower number of samples per speech frame. At the
`decoder, the high frequency contents of the signal needs to
`be reintroduced to remove the low pass filtering effect from
`the decoded synthesized signal and retrieve the natural
`sounding quality of wide band signals. For that purpose, an
`efficient technique for recovering the high frequency content
`60 of the wideband signal is needed to thereby produce a
`full-spectrum wideband synthesized signal, while maintain(cid:173)
`ing a quality close to the original signal.
`
`OBJECT OF THE INVENTION
`
`An object of the present invention is therefore to provide
`such an efficient high frequency content recovery technique.
`
`Page 7 of 20
`
`
`
`3
`SillllMARY OF THE INVENTION
`
`US 7,151,802 B1
`
`4
`wideband signal for prod11cing an over-san1pled signal ver(cid:173)
`sion of tbe synthesized wideband signal; and
`f) a high-frequency content recovering device as
`described hereinabove, for recovering a high frequency
`content of the wideband signal and for injecting the high
`11-equency content in the over-sampled signal version to
`produce the full-spt.'Ctrum synthesized wideband signal.
`In accordance with a preferred embodiment, the decoder
`10 further comprises:
`a) a voicing factor generator responsive to the adaptive
`and innovative codevectors for calculating a voicing factor
`for forwarding to the gain adjustment module;
`b) an energy computing module responsive to the exci(cid:173)
`tation signal for calculating an excitation energy for for(cid:173)
`warding to the gain adjustment module; and
`c) a spectral tilt calculator responsive to the synthesized
`signal for calculating a tilt scaling factor for forwarding to
`the gain adjustment module. 111e first subset of the shaping
`parameters comprises the voicing factor, the energy scaling
`factor, and the tilt scaling factor, and the second subset of the
`shaping parameters includes linear prediction coeftlcients.
`In accordance with other prefen·ed embodiments of the
`decoder:
`the voicing factor generator calculates the voicing factor
`rv using the relation:
`
`More specifically, in accordance with the present inven(cid:173)
`tion, there is provided a method for recovering a high
`frequency content of a wideband signal previously down- 5
`sampled and for injecting the high frequency content in an
`over-sampled synthesized version of the wideband signal to
`produce a full-spectrum synthesized wideband signal. This
`high-frequency content recovering method comprises: gen(cid:173)
`enlting a noise sequence; spectrally-shaping the noise
`sequence in relation to shaping paran1eters representative of
`the down-sampled wideband signal; and injecting the spec(cid:173)
`trally-shaped noise sequence in the over-sampled synthe(cid:173)
`sized signal version to thereby produce the ftdl-spectrum 15
`synthesized wideband signaL
`The present invention :fi.lrther relates to a device for
`recovering a high frequency content of a wideband signal
`previously down-sampled and for injecting this high fre(cid:173)
`quency content in an over-sampled synthesized version of 20
`the wideband signal to produce a full-spectrum synthesized
`wideband signal. This high-frequency content recovering
`device comprises a noise generator for producing a noise
`sequence, a spectral shaping unit tor shaping the noise 25
`sequence in relation to shaping parameters representative of
`the down-sampled wideband signal, and a signal injection
`circuit for injecting the spectrally-shaped noise sequence in
`the over-sampled synthesized signal version to thereby
`p.roduce the full-spectn11n synthesized wideband signaL
`ln accordance with a preferred embodiment, the noise
`sequence is a white noise sequence.
`Preferably, spectral shaping of the noise sequence com(cid:173)
`prises: producing a scaled white noise sequence in response 35
`to the white noise sequence and a first subset of the shaping
`parameters; filtering the scaled white noise sequence in
`relation to a second subset of the shaping parameters com(cid:173)
`pris.ing bandwidth expanded synthesis filter coefficients to
`produce a filtered scaled white noise sequence characterized 40
`by a frequency bandwidth generally higher than a frequency
`bandwidth of the over-sampled synthesized sigr1al version;
`and band-pass .filtering the filtered scaled white noise
`sequence to produce a band-pass filtered scaled white noise
`sequeDce to be subsequently injected in the over-sampled 45
`sy1~thesized signal version as the spectrally-shaped white
`no1se sequence.
`Still according to the present invention, there is provided
`a decoder for producing a synthesized wideband signal,
`compnsmg:
`a) a signal fragmenting device for receiving an encoded
`version of a wideband signal previously down-sampled
`during encoding and extracting from the encoded wideband
`signal version at least pitch codebook parameters, innova(cid:173)
`tive codebook parameters, and synthesis filter coefficients; 55
`b) a pitch codebook responsive to the pitch codebook
`parameters for producing a pitch codevector,
`c) an innovative codebook responsive to the innovative
`code book parameters for producing an innovative codevec(cid:173)
`tor;
`d) a combiner circuit for combining the pitch codevector
`and the innovative codevector to thereby produce an exci(cid:173)
`tation signal;
`e) a signal synthesis device including a synthesis filter for
`filtering the excitation signal in relation to the synthesis filter
`coefficients to thereby produce a synthesized wideband
`signal, and au oversampler responsive to the synthesized
`
`
`30
`
`50
`
`60
`
`where E., is the energy of the gain scaled pitch codevector
`and E.,. is the energy of the gain scaled innovative codevec(cid:173)
`tor;
`the gain a~justing unit calculates an energy scaling factor
`using the relation:
`
`Energy s.:aling factor' =
`
`l'/-J
`E u'1(n)
`n=O
`N'-1
`l: "'2(11)
`'1=0
`
`n=O, . .. , N'-L
`where w' is the white noise sequence and u' is an enhanced
`excitation signal derived from the excitation signal;
`tb.e spectral tilt calculator calculates tb.e tilt scaling factor
`g, using the relation:
`
`g,- 1-tilt bow1ded by 0.2§ g,§ J.O
`
`where
`
`N- 1
`Ls,(n)s,(n -1)
`tilt = .:::"=::.•--,--,---(cid:173)
`"'"' E ~(n)
`•-0
`
`conditioned by tilt~O and tilt~rv.
`65 or tbe relation:
`
`g,=to-<>MI• bounded by 0.2 ;;ig,;;il.O
`
`Page 8 of 20
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`US 7,151,802 B1
`
`6
`DETAILED DESCRIPTION OF THE
`PREFERRED EMBODIMENT
`
`where
`
`5
`
`N-1
`~ sh(n)sh(n- 1)
`tilt= "-n~-'-1---
`N-1
`2.: sl,(n)
`n=O
`
`As well known to those of ordinary skill in the art, a
`cellular communication system such as 401 (see FIG. 4)
`provides a telecommunication service over a large geo(cid:173)
`graphic area by dividing that large geographic area into a
`number C of smaller cells. The C smaller cells are serviced
`by respective cellular base stations 402 1 , 4022 . . . 402c to
`10 provide each cell with radio signalling, audio and data
`channels.
`Radio signalling channels are used to page mobile radio(cid:173)
`telephones (mobile transmitter/receiver units) such as 403
`within the limits of the coverage area (cell) of the cellular
`15 base station 402, and to place calls to other radiotelephones
`403 located either inside or outside the base station's cell or
`to another network such as the Public Switched Telephone
`Network (PSTN) 404.
`Once a radiotelephone 403 has successfully placed or
`20 received a call, an audio or data channel is established
`between this radiotelephone 403 and the cellular base station
`402 corresponding to the cell in which the radiotelephone
`403 is situated, and communication between the base station
`402 and radiotelephone 403 is conducted over that audio or
`data channel. The radiotelephone 403 may also receive
`control or timing information over a signalling channel
`while a call is in progress.
`If a radiotelephone 403 leaves a cell and enters another
`adjacent cell while a call is in progress, the radiotelephone
`30 403 hands over the call to an available audio or data channel
`of the new cell base station 402. If a radiotelephone 403
`leaves a cell and enters another adjacent cell while no call is
`in progress, the radiotelephone 403 sends a control message
`over the signalling channel to log into the base station 402
`35 of the new cell. In this manner mobile communication over
`a wide geographical area is possible.
`The cellular communication system 401 further com(cid:173)
`prises a control terminal 405 to control communication
`between the cellular base stations 402 and the PSTN 404, for
`40 example during a communication between a radiotelephone
`403 and the PSTN 404, or between a radiotelephone 403
`located in a first cell and a radiotelephone 403 situated in a
`second cell.
`Of course, a bidirectional wireless radio communication
`subsystem is required to establish an audio or data channel
`between a base station 402 of one cell and a radiotelephone
`403 located in that cell. As illustrated in very simplified form
`in FIG. 4, such a bidirectional wireless radio communication
`subsystem typically comprises in the radiotelephone 403:
`a transmitter 406 including:
`an encoder 407 for encoding the voice signal; and
`a transmission circuit 408 for transmitting the encoded
`voice signal from the encoder 407 through an antenna
`such as 409; and
`a receiver 410 including:
`a receiving circuit 411 for receiving a transmitted encoded
`voice signal usually through the same antenna 409; and
`a decoder 412 for decoding the received encoded voice
`signal from the receiving circuit 411.
`The radiotelephone further comprises other conventional
`radiotelephone circuits 413 to which the encoder 407 and
`decoder 412 are connected and for processing signals there(cid:173)
`from, which circuits 413 are well known to those of ordinary
`skill in the art and, accordingly, will not be further described
`in the present specification.
`Also, such a bidirectional wireless radio communication
`subsystem typically comprises in the base station 402:
`
`conditioned by tilt~O and tilt~rv.
`Preferably, the band-pass filter has a frequency bandwidth
`located between 5.6 kHz and 7.2 kHz.
`Also according to the present invention, in a decoder for
`producing a synthesized wideband signal, comprising:
`a) a signal fragmenting device for receiving an encoded
`version of a wideband signal previously down-sampled
`during encoding and extracting from the encoded wide band
`signal version at least pitch codebook parameters, innova(cid:173)
`tive codebook parameters, and synthesis filter coefficients;
`b) a pitch codebook responsive to the pitch codebook
`parameters for producing a pitch codevector;
`c) an innovative codebook responsive to the innovative
`code book parameters for producing an innovative codevec-
`tor;
`d) a combiner circuit for combining the pitch codevector
`and the innovative codevector to thereby produce an exci(cid:173)
`tation signal; and
`e) a signal synthesis device including a synthesis filter for
`filtering the excitation signal in relation to the synthesis filter
`coefficients to thereby produce a synthesized wideband
`signal, and an oversampler responsive to the synthesized
`wideband signal for producing an over-sampled signal ver(cid:173)
`sion of the synthesized wideband signal;
`
`25
`
`the improvement comprising a high-frequency content
`recovering device as described hereinabove for recovering a
`high frequency content of the wideband signal and for
`injecting the high frequency content in the over-sampled
`signal version to produce the full-spectrum synthesized
`wideband signal.
`The present invention finally comprises a cellular com(cid:173)
`munication system, a cellular mobile transmitter/receiver
`unit, a cellular network element, and a bidirectional wireless 45
`communication sub-system comprising the above described
`decoder.
`The objects, advantages and other features of the present
`invention will become more apparent upon reading of the
`following non restrictive description of a preferred embodi- 50
`ment thereof, given by way of example only with reference
`to the accompanying drawings.
`
`BRIEF DESCRIPTION OF THE DRAWINGS
`
`55
`
`In the appended drawings:
`FIG. 1 is a schematic block diagram of a preferred
`embodiment of wide band encoding device;
`FIG. 2 is a schematic block diagram of a preferred
`embodiment of wide band decoding device;
`FIG. 3 is a schematic block diagram of a preferred
`embodiment of pitch analysis device; and
`FIG. 4 is a simplified, schematic block diagram of a
`cellular communication system in which the wideband 65
`encoding device of FIG. 1 and the wideband decoding
`device of FIG. 2 can be used.
`
`60
`
`Page 9 of 20
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`US 7,151,802 B1
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`15
`
`8
`x' Target vector for innovation search;
`h Weighted synthesis filter impulse response;
`v r Adaptive (pitch) code book vector at delay T;
`y r Filtered pitch code book vector (v r convolved with h);
`ck Innovative codevector at index k (k-th entry from the
`innovation codebook);
`cf Enhanced scaled innovation codevector;
`u Excitation signal (scaled innovation and pitch codevec-
`tors );
`U' Enhanced excitation;
`z Band-pass noise sequence;
`w' White noise sequence; and
`w Scaled noise sequence.
`List of Transmitted Parameters
`STP Short term prediction parameters (defining A(z));
`T Pitch lag (or pitch codebook index);
`b Pitch gain (or pitch codebook gain);
`j Index of the low-pass filter used on the pitch codevector;
`k Codevector index (innovation codebook entry); and
`g Innovation codebook gain.
`In this preferred embodiment, the STP parameters are
`transmitted once per frame and the rest of the parameters are
`transmitted four times per frame (every subframe).
`Encoder Side
`The sampled speech signal is encoded on a block by block
`basis by the encoding device 100 of FIG. 1 which is broken
`down into eleven modules numbered from 101 to 111.
`The input speech is processed into the above mentioned
`L-sample blocks called frames.
`Referring to FIG. 1, the sampled input speech signal114
`is down-sampled in a down-sampling module 101. For
`example, the signal is down-sampled from 16 kHz down to
`12.8 kHz, using techniques well known to those of ordinary
`35 skill in the art. Down-sampling down to another frequency
`can of course be envisaged. Down-sampling increases the
`coding efficiency, since a smaller frequency bandwidth is
`encoded. This also reduces the algorithmic complexity since
`the number of samples in a frame is decreased. The use of
`down-sampling becomes significant when the bit rate is
`reduced below 16 kbit/s, although down-sampling is not
`essential above 16 kbit/s.
`After down-sampling, the 320-sample frame of 20 ms is
`reduced to 256-sample frame (down-sampling ratio of 'Ys ).
`The input frame is then supplied to the optional pre(cid:173)
`processing block 102. Pre-processing block 102 may consist
`of a high-pass filter with a 50 Hz cut-off frequency. High(cid:173)
`pass filter 102 removes the unwanted sound components
`below 50 Hz.
`The down-sampled pre-processed signal is denoted by
`sp(n), n=O, 1, 2, ... , L-1, where Lis the length of the frame
`(256 at a sampling frequency of 12.8 kHz). In a preferred
`embodiment of the preemphasis filter 103, the signal sP(n) is
`preemphasized using a filter having the following transfer
`function:
`
`7
`a transmitter 414 including:
`an encoder 415 for encoding the voice signal; and
`a transmission circuit 416 for transmitting the encoded
`voice signal from the encoder 415 through an antenna
`such as 417; and
`a receiver 418 including:
`a receiving circuit 419 for receiving a transmitted encoded
`voice signal through the same antenna 417 or through
`another antenna (not shown); and
`a decoder 420 for decoding the received encoded voice 10
`signal from the receiving circuit 419.
`The base station 402 further comprises, typically, a base
`station controller 421, along with its associated database
`422, for controlling communication between the control
`terminal 405 and the transmitter 414 and receiver 418.
`As well known to those of ordinary skill in the art, voice
`encoding is required in order to reduce the bandwidth
`necessary to transmit sound signal, for example voice signal
`such as speech, across the bidirectional wireless radio com(cid:173)
`munication subsystem, i.e., between a radiotelephone 403 20
`and a base station 402.
`LP voice encoders (such as 415 and 407) typically oper(cid:173)
`ating at 13 kbits/second and below such as Code-Excited
`Linear Prediction (CELP) encoders typically use a LP syn(cid:173)
`thesis filter to model the short-term spectral envelope of the 25
`voice signal. The LP information is transmitted, typically,
`every 10 or 20 ms to the decoder (such 420 and 412) and is
`extracted at the decoder end.
`The novel techniques disclosed in the present specifica(cid:173)
`tion may apply to different LP-based coding systems. How- 30
`ever, a CELP-type coding system is used in the preferred
`embodiment for the purpose of presenting a non-limitative
`illustration of these techniques. In the same manner, such
`techniques can be used with sound signals other than voice
`and speech as well with other types of wideband signals.
`FIG. 1 shows a general block diagram of a CELP-type
`speech encoding device 100 modified to better accommo(cid:173)
`date wideband signals.
`The sampled input speech signal 114 is divided into
`successive L-sample blocks called "frames". In each frame, 40
`different parameters representing the speech signal in the
`frame are computed, encoded, and transmitted. LP param(cid:173)
`eters representing the LP synthesis filter are usually com(cid:173)
`puted once every frame. The frame is further divided into
`smaller blocks of N samples (blocks of length N), in which 45
`excitation parameters (pitch and innovation) are determined.
`In the CELP literature, these blocks of length N are called
`"subframes" and theN-sample signals in the subframes are
`referred to as N-dimensional vectors. In this preferred
`embodiment, the length N corresponds to 5 ms while the 50
`length L corresponds to 20 ms, which means that a frame
`contains four subframes (N=80 at the sampling rate of 16
`kHz and 64 after down-sampling to 12.8 kHz). Various
`N-dimensional vectors occur in the encoding procedure. A
`list of the vectors which appear in FIGS. 1 and 2 as well as 55
`a list of transmitted parameters are given herein below:
`List of the Main N-Dimensional Vectors
`s Wideband signal input speech vector (after down-
`sampling, pre-processing, and preemphasis );
`sw Weighted speech vector;
`s0 Zero-input response of weighted synthesis filter;
`sP Down-sampled pre-processed signal;
`Oversampled synthesized speech signal;
`s' Synthesis signal before deemphasis;
`sd Deemphasized synthesis signal;
`sh Synthesis signal after deemphasis and postprocessing;
`x Target vector for pitch search;
`
`P(z)~1-!1Z-I
`
`where fl is a preemphasis factor with a value located between
`60 0 and 1 (a typical value is fl=0.7). A higher-order filter could
`also be used. It should be pointed out that high-pass filter
`102 and preemphasis filter 103 can be interchanged to obtain
`more efficient fixed-point implementations.
`The function of the preemphasis filter 103 is to