JP2000206998A - Receiver and receiving method, communication equipment and communicating method - Google Patents

Abstract

PROBLEM TO BE SOLVED: To provide a receiver and receiving method, communication equipment and communicating method in which hearing quality of received voices is improved in accordance with user's desire not only during a real time operation but also during the reproducing of an automatic recording. SOLUTION: During an automatic recording, voice parameter codes having a sampling frequency of 8 kHz are recorded in a memory 13. During a reproducing of the automatic recording, a signal processing device 5 reads the codes from the memory 13, reproduces first band voice signals having sampling frequencies of 8 kHz and 16 kHz and wide band voice signals having a sampling frequency of 16 kHz and outputs these signals by switching them. Thus, in a receiver 1, the recording capacity of the memory 13 is saved and high quality voices are reproduced based on the user's desire.

Description

DETAILED DESCRIPTION OF THE INVENTION

[0001]

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to a receiving apparatus and method for synthesizing an audio signal using an audio parameter code of an audio signal transmitted by communication or broadcasting, and a communication apparatus and method.

[0002]

2. Description of the Related Art In a conventional communication device, a sampling frequency of an input voice and an output voice on a receiving side are the same, and a voice frequency band is also the same. This is because the transmission band of the telephone line is narrow, for example, 300 to 3400 Hz, and the frequency band of the audio signal transmitted via the telephone line is limited.

Some of the above communication devices have a recording medium such as a semiconductor memory on the receiving side, and have a so-called answering machine recording function of recording parameters of a transmitted voice signal and reproducing the parameters later. However, during reproduction, the sampling frequency and the frequency band are kept as they are.

[0004]

By the way, the sound quality of the sound output in the same sound frequency band as the input sound whose transmission band is limited is not so good. That is, the auditory quality is inferior. They also complain about the sound quality of digital mobile phones.

[0005] The present invention has been made in view of the above-mentioned circumstances, and it is possible to obtain a received voice with improved auditory quality as desired by a user not only during real-time operation but also during voice mail playback. It is an object of the present invention to provide a receiving device and method, a communication device and a method.

[0006]

In order to solve the above-mentioned problems, a receiving apparatus according to the present invention includes a transmitting signal transmitted from a transmitting apparatus to generate an audio signal having a first sampling frequency _fs1. A recording medium for recording an audio parameter code based on the first audio signal, and a sampling frequency of an audio signal of a _first band B1 generated using the audio parameter code recorded on the recording medium.
_s2 ( _fs2 > _fs1 ), and a second rate component out of the _first band B1 using the audio parameter code recorded on the recording medium.
And out-of-band component predicting unit to estimate the audio signal of the second sampling frequency f _s2 of band B ₂ of the audio signal and the second sampling frequency f _s2 at the sampling rate converting means, above the audio signal and a switching means for switching between audio signals of the second second broadband B _W obtained by adding the audio signal of the band B ₂ of the sampling frequency f _s2, which was estimated by band components estimating means.

Here, the switching means also switches the audio signal of the first band B ₁ of the first sampling frequency f _s1 generated using the audio parameter code recorded on the recording medium.

In order to solve the above-mentioned problems, a receiving method according to the present invention reads out a voice parameter code based on a transmission signal transmitted to generate a voice signal of a first sampling frequency f _s1 from a recording medium. Generated first
Of the conversion output of the sampling frequency of the audio signal obtained by converting the second sampling frequency _{f s2 (f s2> f s1} ) of the band B _1, the speech parameter codes read from the recording medium to the converted output First band B estimated using
Switching between the sum output plus second audio signal of a second sampling frequency f _s2 of band B ₂ is out-of-band component of the _1.

[0009] Here, the receiving method switches also first band B ₁ of the audio signal of a first sampling frequency f _s1 generated using a speech parameter codes read from the recording medium.

[0010] In order to solve the above-mentioned problems, a communication device according to the present invention includes: a transmitting unit that performs an encoding process on an input audio signal at a first sampling frequency f _s1 to generate a transmission signal; The sampling frequency of the audio signal of the first band B1 generated by reading out the audio parameter code based on the transmission signal generated by performing the encoding process at the sampling frequency f _s1 from the recording medium is changed to the second sampling frequency f _s2 (f _s2> f _s1) and the conversion obtained was converted output, the second of the second band B ₂ of the audio sampling frequency f _s2 of guessed using the speech parameter codes read from the recording medium to the converted output Receiving means for switching between an added output obtained by adding the signals.

Here, the receiving means includes: a recording medium for recording an audio parameter code based on a transmission signal transmitted from a transmitting device to generate the audio signal of the first sampling frequency f _s1 ; the parameter codes and sampling rate conversion means for converting the sampling frequency of the first band B ₁ of the audio signal generated by reading from the recording medium to the second sampling frequency _{f s2 (f s2> f s1} ), said recording medium and out-of-band component predicting unit for using the speech parameter codes to infer an audio signal of the second sampling frequency f _s2 of the second band B ₂ is a band component of the first band B ₁ read from, The second sampling frequency f _{s2 is} calculated by the sampling rate conversion means.
And audio signal and a second second broadband B _W of the audio signal the audio signal having a bandwidth of B ₂ obtained by adding the sampling frequency f _s2, which was estimated by the band components estimating means to the audio signal And switching means for switching between.

The switching means also switches the audio signal of the first band B ₁ of the first sampling frequency f _s1 generated using the audio parameter code.

In order to solve the above-mentioned problems, a communication method according to the present invention performs an encoding process on an input audio signal at a first sampling frequency f _s1 to generate a transmission signal, and generates a transmission signal. The first parameter read out from the recording medium and generated based on the voice parameter code based on the transmission signal transmitted to generate the voice signal of f _s1 .
The second sampling frequency of the band B ₂ is out-of-band component of the band B ₁ of the second sampling frequency f _s2 (f _s2> f
switching the converted output obtained by converting the _s1), and the converted output to the addition obtained by adding the second audio signal having a bandwidth B ₂ of the second sampling frequency f _s2 of guessed using the speech parameter codes outputted .

[0014] In the communication method, the first communication method may include a first parameter generated using a voice parameter code read from the recording medium.
Also switching the first band B ₁ of the audio signal having the sampling frequency f _s1.

[0015]

Embodiments of the present invention will be described below with reference to the drawings. This embodiment is a receiving apparatus 1 shown in FIG. 1 which is a specific example of a receiving apparatus according to the present invention, to which a receiving method according to the present invention is applied. This receiving device 1 is a personal digital cellular (Pers
onal Digital Cellular (PDC), which can be used as a receiving side of a digital mobile phone that is currently widely used.

The receiving apparatus 1, from the speech parameter code transmitted via a base station from the transmitting apparatus to be described later to generate an audio signal of a first sampling frequency f _s1, the first sampling frequency f _s1 The audio signal of the first band B ₁ and the second sampling frequency f _s2 (f _s2 > f
a first audio signal having a bandwidth B _{1 s1),} a second sampling frequency _{f s2 (f s2> f s1} ) wideband B _w (first band B ₁ + second band B ₂₎ of the speech signal Is generated, and these three types of audio signals are switched and output. 8 KHz is used as the first sampling frequency f _{s1 and} 16 KHz is used as the second sampling frequency f _s2 . As the first band B ₁ of 300Hz～3400Hz, using 3400Hz~6000Hz the second as band B _2. Therefore, the wide-band B _W 300Hz~60
Assume 00 Hz.

The receiving device 1 has a so-called answering machine recording function, and stores the voice parameter code in the semiconductor memory 13 and stores the voice parameter code in the memory 13 in response to a user operation. And read
The transmitted voice transmitted during absence can be switched and output as the above three types of voice signals.

The voice parameter code received from the base station via the antenna 2 is stored in the memory 5a of the signal processing device 5 via the RF receiver 3 and the controller 4. When the answering machine recording function is set by the user, the voice parameter code is stored in the memory 13.

The speech parameter code stored in the memory 5a of the signal processing device 5 is decoded by the decoding unit of the signal processing device 5, then subjected to predetermined signal processing and output.
Further, when reproducing the transmitted voice recorded by the answering machine by the user's operation, the voice parameter code stored in the memory 13 is decoded by the decoding unit of the signal processing device 5 and then subjected to predetermined signal processing. Is output. Hereinafter, the former is referred to as a real-time operation, and the latter is referred to as an answering machine reproduction.

At the time of real-time operation and answering machine reproduction, the output signal from the signal processor 5 is converted into an analog signal by the D / A converter 6 and then passed through the anti-aliasing filter 7, volume 8 and amplifier 9. Speaker 10
Output from The control unit 4 is connected to, for example, a key operation unit 11 and an LCD display unit 12.

FIG. 2 shows a configuration of a transmitting device 15 for transmitting the above-mentioned voice parameter code via, for example, a radio transmission path and a base station.

The audio signal input from the microphone 16 is stored in the memory 21a of the signal processing device 21 via the amplifier 17, the volume 18, the anti-aliasing filter 19, and the A / D converter 20.

The audio signal stored in the memory 21a is subjected to code processing in an audio encoding unit inside the signal processing device 21, and is output as an audio parameter code. This voice parameter code is transmitted to the control unit 22 and the RF (RF transmission) amplifier 23
And transmitted to the base station via the antenna 24. The control unit 22 includes a key operation unit 25 and an LCD display unit 26.
Is connected.

Here, the voice coding unit inside the signal processing device 21 generates a voice parameter code in consideration of narrowing of a band limited by a wireless transmission path. Generally, 300
A transmission band of 3 Hz to 3400 Hz is considered. The voice parameter code is supplied to the RF amplifier 23 via the control unit 22. Then, the signal is transmitted from the antenna 24 as an RF signal.

The voice parameter code transmitted from the transmitting device 15 via the base station is recorded in the memory 13 of the receiving device 1 when the answering machine recording function is set as described above.

Table 1 shows the digital telephone standards.
7 shows audio parameter codes in an example of a one-frame full-rate audio data format. Table 2 shows audio parameter codes in a half-rate format example.

[0027]

[Table 1]

[0028]

[Table 2]

The speech parameter codes used in the present invention include a linear prediction (LPC) residual for the excitation source, a linear prediction coefficient α, a lag LAG for the pitch frequency as shown in Tables 1 and 2, and, for example, There are frame power R0, CRC and the like in a frame of 20 msec. These speech parameter codes have a 10-word structure in which 16 bits (bit) are defined as 1 word (Word) per frame.
That is, it is transmitted in 160 samples.

At the time of the answering machine reproduction, the voice parameter code is read from the memory 13 and sent to the signal processing device 5 inside the receiving device 1.

The signal processing device 5 inside the receiving device 1 of FIG.
Comprises a decoder 27 shown in FIG. 3 and a signal switching unit 32 shown in FIG.

The encoding method in the speech encoding unit in the signal processing device 21 of the transmitting device 15 shown in FIG.
If it is based on the CELP (Pitch Synchronous Innovation-CELP) coding method, the decoder 27 decodes the sound using the transmission signal based on the PSI-CELP coding, and outputs an output terminal 28. supplying an excitation source NExc _N decoded audio Snd _N, the linear prediction coefficient alpha _N to the output terminal 29, an output terminal 30 to.

The signal switching section 32 uses the PSI-CELP code transmitted from the transmitting apparatus to generate the audio signal of the first sampling frequency f _s1 (= 8 KHz), and the signal is decoded by the decoder 27. First band B
₁ (300Hz-3400Hz) decoded sound Snd _N
Of the second sampling frequency f _s2
(= 16 KHz), a linear prediction coefficient α _N obtained by the decoder 27 decoding the PSI-CELP code, and an excitation source NExc _N.
And the second sampling frequency f _s2 (= 16 KH
second band of z) B ₂ (3400Hz~6000Hz)
Out of band component estimating means for estimating the signal of the first sampling frequency f _s1 (= 8 KHz) and the first band B ₁ (3
00 Hz to 3400 Hz) and a first band B of a second sampling frequency f _s2 (= 16 KHz).
₁ (300 Hz to 3400 Hz) and a wideband B _{w at} the second sampling frequency f _s2 (= 16 KHz)
(300 Hz to 6000 Hz) audio signal switching means for switching between the three types of audio signals for output.

Here, the second sampling frequency f
_s2 (= 16KHz) of broadband B _w (300Hz~600
0 Hz) is estimated by the out-of-band component estimating means and the decoded audio Snd _{N of} the first band B ₁ (300 Hz to 3400 Hz), which is set to the second sampling frequency f _s2 by the sampling rate converting means. The second band B of the obtained second sampling frequency f _s2
2 (3400 Hz to 6000 Hz) is obtained by adding the signals by an adder 46 serving as an adding means.

The above-mentioned sampling rate conversion means is the up-sampling section 45 in FIG. The switching means is a changeover switch 47. The out-of-band component estimating means is a part excluding the up-sampling unit 45, the adding unit 46, and the changeover switch 47.

Hereinafter, the configuration of the signal switching section 32 will be described in detail. First, the out-of-band component estimating means will be described. The out-of-band component estimating means includes a linear prediction coefficient → autocorrelation (α _N → r _N ) conversion unit 36, an autocorrelation (r) widening unit 37, a wideband codebook (r _w CB) 38, → Linear prediction coefficient (r _w → α _w ) converter 39 and LP
It comprises a C synthesizing unit 40, an excitation source expanding unit 41, a high-frequency extraction and suppression filter 42, and a multiplying unit 43.

The linear prediction coefficient α _N supplied from the input terminal 34 is converted from the linear prediction coefficient → autocorrelation (α _N → r _N ) conversion unit 3
6. The α _N → r _N conversion unit 36 converts the linear prediction coefficient α _N into an autocorrelation r _N , and supplies the auto correlation r _N to the autocorrelation (r) widening unit 37. The autocorrelation (r) widening unit 37 widens (extends) the autocorrelation r using the wideband codebook (r _w CB) 38. Wideband codebook (r
_w CB) 38 is created in advance using the autocorrelation parameter r _w extracted from the wideband sound.

[0038] Using the wide band code book (r _w CB) 38, extended autocorrelation r _w autocorrelation (r) broadband portion 37 is expanded in the autocorrelation → linear prediction coefficients (r _w → α _w) conversion unit 39 Supplied. The r _w → α _w conversion unit 39 converts the extended auto-correlation r _w into the extended linear prediction coefficient α _w again, and supplies it to the LPC synthesis unit 40. LPC synthesis section 40 synthesizes a wideband speech based on the extended excitation source from the excitation source extension 41 to be described later with r _w → _{α w} wideband linear prediction coefficients from the conversion unit 39 alpha _w.

The combined output of the LPC combining section 40 is supplied to a high-frequency extraction and suppression filter 42. The high-frequency extraction & suppression filter 42 removes signal components in the frequency band of 300 Hz to 3400 Hz, and the second band B ₂ = 3400 Hz to 60
High frequency components are suppressed so that 00 Hz signal components are extracted. The filter output from the filter 42 is multiplied by a gain supplied from a terminal 44 by a multiplier 43. The output (second band B ₂ = 3400 Hz to 6000 Hz) multiplied by the gain in the multiplier 43 is output to the adder 4
6.

The extended excitation source from the excitation source extension unit 41 is also supplied to the LPC synthesis unit 40 as described above. Excitation source extension 41 extends the LPC residuals of the parameters relating to an excitation source supplied from an input terminal 35 (the LPC residuals referred to as the excitation source NExc _N.). FIG. 5 shows a detailed configuration of the excitation source extension unit 41.

First, the excitation source NExc _N supplied through the input terminal 35 is up-sampled by the up-sampling section 50. The output of the up-sampling unit 50 is LP
F51, LP from output terminal 55 via boost section 52
It is sent to the C synthesizing unit 40. That is, the excitation source NExc _N
Is used as the above-mentioned extended excitation source when synthesizing the audio signal. The boost unit 52
This is for boosting the extended excitation source when an affricate or fricative is detected, and the boost amount is controlled by an output of the affricate detector 54. Affricate detector 54
Receives the autocorrelation r _N from the α _N → r _N conversion unit 36 via the input terminal 53 and detects affricate and fricative.

The signal switching section 32 serves as the sampling rate conversion means as described above,
Supplied from the first band B ₁ = 300~3400H
The sampling frequency of the decoded sound Snd _N of z is f _s1
And an up-sampling section 45 for up-sampling from 8 kHz to f _s2 = 16 kHz.

Then, in the up-sampling section 45, the sampling frequency is set to the second sampling frequency f _s2 = 16 kHz.
converted to z, first band B ₁ = 300 Hz to 340
The adder 46 adds the audio signal component of 0 Hz and the audio signal component of the second band B ₂ = 3400 Hz to 6000 Hz of the second sampling frequency f _s2 = 16 kHz, which is the multiplied output from the multiplier 43.

Further, as described above, the signal switching section 32 includes the switching switch 47 as the switching means, and the first band B ₁ (300 Hz to 3400 Hz) of the first sampling frequency f _s1 (= 8 KHz). Audio signal and
The first of the second sampling frequency f _s2 (= 16 KHz)
Is switched between the audio signal of the band B ₁ (300 Hz to 3400 Hz) and the audio signal of the wide band B _w (300 Hz to 6000 Hz) of the second sampling frequency f _s2 (= 16 KHz).

The changeover switch 47 is connected to the first band B of the first sampling frequency f _s1 (= 8 KHz).
₁ (300 Hz to 3400 Hz) is received by the selected terminal a, and the second sampling frequency f _s2 (= 16K
Hz) first band B ₁ (300 Hz to 3400 Hz)
Receiving a voice signal at a selected terminal b, broadband B _w (300 Hz of the second sampling frequency f _s2 (= 16KHz)
To 6000 Hz) at the selected terminal c.
Then, by switching the selection piece d based on the switching control signal from the switching control signal terminal 49, one of the audio signals is supplied to the D / A converter 6.

The principle of operation of the signal switching section 32 having the above configuration will be described below. Signal switching unit 32
Is 3400 from a speech parameter code for generating a speech signal of a _first band B1 of 300 Hz to 3400 Hz.
A speech coding parameter for a _second band B2 of ₂ Hz to 6000 Hz is generated, and wideband LPC synthesis is performed. Then, the low frequency band (300 Hz to 34 Hz)
00 Hz) side is replaced with the original sound up-sampled to 16 KHz. That is, a high-pass filter is applied to leave only the high band (3400 Hz to 6000 Hz), high frequency components among these high band components are suppressed, and the gain is further adjusted.
Hz) is up-sampled (second sampling frequency f
_s2 ) and add the second sampling frequency f
_s2 (= 16KHz) of broadband B _w (300Hz~600
0 Hz).

Here, to widen (or expand) the speech parameter code, it is necessary to broaden the linear prediction coefficient α and the excitation source NExc _N. Also, α
In order to widen the bandwidth, it is necessary to previously create a codebook based on autocorrelation r, which is a parameter that can be mutually converted with α. The band of the autocorrelation r is widened by quantization and inverse quantization by the codebook.

First, the widening of the linear prediction coefficient α will be described. Note that α is a filter coefficient representing a spectral envelope, and is temporarily converted into an autocorrelation r, which is a parameter representing another spectral envelope that makes it easy to estimate the high frequency side,
This is widened, and then inversely converted from the wideband (or extended) autocorrelation r _w to the wideband (or extended) linear prediction coefficient α _w . Vector quantization is used for extension. Narrowband autocorrelation r _n to vector quantization, may be obtained the corresponding r _w from that index.

Since a certain relationship holds between the narrow-band autocorrelation and the wideband autocorrelation as described later, only a codebook based on the wideband autocorrelation need be prepared, and the narrowband autocorrelation can be vector-quantized by this. Wideband autocorrelation is obtained by inverse quantization.

Assuming that the narrow-band signal is obtained by band-limiting the wide-band signal, the wide-band auto-correlation and the narrow-band auto-correlation have a relationship represented by the following equation (1).

[0051]

(Equation 1)

Here, φ is an autocorrelation, _xn is a narrow band signal, _xw is a wide band signal, and h is an impulse response of a band limiting filter.

Further, the following equation (2) is obtained from the relationship between the autocorrelation and the power spectrum.

[0054]

(Equation 2)

If another band-limiting filter having a frequency characteristic equal to the power characteristic of this band-limiting filter is considered and this is set to H ′, the above equation (2) can be expressed by the following equation (3).
It looks like an expression.

[0056]

(Equation 3)

The pass band and the stop band of this new filter are the same as those of the original band limiting filter, and the attenuation characteristic is squared. Therefore, this new filter can also be said to be a band limiting filter. With this in mind, narrowband autocorrelation is simplified to the convolution of broadband autocorrelation with the impulse response of a band-limited filter, ie, band-limited wideband autocorrelation. That is, the following equation (4) is obtained.

[0058]

(Equation 4)

As described above, when performing the vector quantization of the narrow-band autocorrelation, if only a wide-band codebook is prepared, the narrow-band vector required at the time of quantization can be created by calculation. It turns out that there is no need to prepare a codebook.

[0060] Further, in order to have a curve r _w code vector of each wide-band autocorrelation increase or decrease monotonically decreasing or gradually, the H 'by no significant change be passed through a low-pass, r _n quantization directly It can be carried out in the r _w code book.
However, since the sampling frequency is 1/2, it is necessary to compare every other order.

The linear prediction coefficient α is expanded because voice data (V) and unvoiced sound (UV) can be further accurately expanded by dividing the voice data into unvoiced sounds (UV). Accordingly, two codebooks for V and UV are used.

Next, expansion of the excitation source will be described. P
In the SI-CELP, an excitation source in a narrow band is upsampled by inserting a zero value in an upsampling unit 50 in FIG. 5 to generate an aliasing distortion. Although this method is very simple, it can be said that the quality is sufficient as an excitation source because the difference between the power and the harmonic structure of the original voice is preserved.

Then, LPC combining is performed by the LPC combining section 40 using the broadband α and the broadband excitation source obtained above.

Since the quality of the wideband LPC-synthesized voice is poor as it is, the original voice Snd _N of the codec output is replaced on the low-frequency side. For this purpose, 3400 Hz or more is extracted from the synthesized sound, while the codec output is up-sampled to fs = 16 KHz, and these are added.

At this time, the gain to be multiplied on the high frequency side by the multiplier 43 can be adjusted by the gain adjuster according to the user's preference. This value is variable because individual differences between users are large. The value of the high-frequency gain is set in advance by an input from the user, and multiplication is performed with reference to this value.

Before addition, high-frequency extraction &
By applying a filtering that slightly suppresses a component of about 6 KHz or more by the suppression filter 42, a sound that is easy to hear is obtained. This filter coefficient is selectable, and processing is performed using a filter selected in advance, so that a higher frequency band can be selected as desired. The selection of this filter is also set by the user's input.

The processing using the filter 42 does not affect the power characteristics on the low frequency side, and may be performed after the addition. Alternatively, it is also possible to apply a filter that also affects the low-frequency side after addition. As described above, a wideband sound can be obtained.

Next, based on the above operation principle, the signal switching section 32 sets the second sampling frequency f _s2 (= 16 KH).
The process of generating an audio signal of the broadband B _w (300Hz~6000Hz) of z) will be described with reference to the flowchart of FIG.

In step S1, the α _N → r _N conversion section 36 shown in FIG. 4 converts the linear prediction coefficient α _N decoded by the decoder 27 shown in FIG. 3 into an autocorrelation r _N. Also,
The audio signal Snd _N decoded by the decoder 27 is subjected to V / UV determination in step S2.

If the result of the determination in step S2 is V, in step S4 the voiced autocorrelation r _N is quantized. This quantization uses the narrowband V parameter obtained in step S3. That is, the parameters for the narrow band V obtained by comparing every other order from the code book 38 of the wide band V are used.

On the other hand, if the result of the determination in step S2 is UV, in step S4 the autocorrelation r for unvoiced sound is quantized using the narrow-band UV parameters obtained in step S3.

Then, in step S5, the wide band V
Inverse quantization using a codebook or a wideband UV codebook, which results in a wideband autocorrelation r _W. The broadband autocorrelation r _W is calculated in step S 6 by r _W → α _W converter 39
To α _W.

On the other hand, the excitation source from the decoder 27 is up-sampled by padding zeros between the samples by the up-sampling unit 50 shown in FIG. 5 in step S7, and is widened by aliasing. This is supplied to the LPC synthesis section 40 as a broadband excitation source.

Then, in step S8, the LPC synthesizing unit 4
0 indicates that the wideband α _W and the wideband excitation source are LPC-combined to obtain a wideband audio signal.

However, if this is the case, it is merely a wideband signal obtained by prediction, and the quality is poor because it contains errors due to prediction. In particular, regarding the frequency range of the input narrowband audio (300 Hz to 3400 Hz), it is better to use the original audio Snd _N (input audio) output from the codec as it is.

Therefore, of the synthesized sounds from the LPC synthesizing section 40, the frequency range of the input narrowband sound is 300 to 340.
0 Hz is set to the band stop filter (BS
It is removed by filtering using P).

Then, in step S10, the up-sampler 45 up-samples the original sound Snd _N and adds it to the adder 46 in step S13. At this time, by filtering the high-frequency side with a high-frequency extraction and suppression filter 42 that slightly suppresses a component of about 6 KHz or more in step S11, the sound is easy to hear. This filter coefficient can be selected as described above.

Further, in step S12, the multiplication section 43
To adjust the high-frequency gain according to the user's preference.

Here, the signal switching unit 32 uses
The creation of a codebook will be described. The creation of the codebook is generally well-known by GLA (Generalized Lloyd A
lgorithm). Broadband audio for a certain time,
For example, the frame is divided into frames every 20 msec, and the autocorrelation of a certain order, for example, the sixth order is obtained for each frame.
The autocorrelation for each frame is used as training data,
Create a 6-dimensional codebook. At this time, voiced sound,
Unvoiced sounds may be distinguished, and the autocorrelation of voiced sounds and the autocorrelation of unvoiced sounds may be separately collected to create respective codebooks. In this case, the code book is referred to when α is expanded during the band expansion processing. At this time, a voiced sound or an unvoiced sound is determined, and the corresponding code book is used.

The signal switching unit 32 uses a codebook for wideband voiced sound and a codebook for wideband unvoiced sound.
Fig. 7 shows the creation of this codebook for wideband voiced sounds.
Fig. 8 shows how to create a codebook for wideband unvoiced sound.
This will be described with reference to FIG.

First, a wideband audio signal is prepared for learning,
At step S31 in FIG. 7, framing is performed for 20 msec per frame. Next, in step S32, for each frame, classification is performed as to whether it is a voiced sound (V) or an unvoiced sound (UV) by examining, for example, a frame energy, a value of zero crossing, and the like.

Then, in step S33, for example, the autocorrelation parameter r up to the sixth order is calculated in the wideband voiced sound frame. In step S34, for example, the autocorrelation parameter r up to the sixth order in the wideband unvoiced sound frame is obtained.

A wideband parameter is extracted from the sixth-order autocorrelation parameters of each frame in step S41 of FIG. 8, and a wideband V (UV) codebook of dimension 6 is created by GLA in step S42.

As described above, in the signal switching unit 32 according to the decoding method based on PSI-CELP, the first band B having a sampling frequency of 8 KHz is set based on the user's preference.
Can send _first audio signal (ranging from 300 to 3400 Hz), the D / A converter 6 sampling frequency is first audio signal or the sampling frequency of the band B ₁ of 16KHz switching the audio signal of the wide band B _W of 16KHz .

Further, the voice parameter codes recorded in the memory 13 at the time of answering machine recording are read out at the time of answering machine playback, and the above three types of audio signals can be generated and switched to be output. In particular, the sampling frequency is 8KHz to 16KH
High quality wideband speech converted to z can be generated.

For this reason, the receiver 1 shown in FIG.
Is the P of the _first band B ₁ (300 Hz to 3400 Hz) whose sampling frequency is different from 8 kHz and 16 kHz.
The received voice signal by SI-CELP and a wide band (300 Hz to 6000H) with a sampling frequency of 16 KHz
The received voice signal by the PSI-CELP of z) can be reproduced according to the user's selection at the time of real-time reproduction and the answering machine green reproduction. The user has more options. Depending on the situation, it is possible to not only extend the received voice band but also make it the same as the input band,
The built-in battery can be reduced.

Also, a voice parameter code having a sampling frequency of 8 KHz is recorded in the memory 13, and a wideband voice having a sampling frequency of 16 KHz can be used as reproduced voice. Audio can be played.

The sampling frequency of the D / A converter 6 is fixed at 16 KHz, and the first frequency is fixed at 16 KHz.
And the audio signal having a bandwidth B ₁ in, may be switched to the audio signal of the wide band B _W. Since the clock used in the D / A converter 6 does not need to be switched to 8 kHz / 16 kHz, the hardware load can be reduced.

In the up-sampling section 45, the changeover switch 47 sets the sampling frequency to 8KH.
The filter output is cleared before switching to z / 16 KHz. This is to prevent generation of noise.

Next, another specific example of the signal processing device 5 in the receiving device 1 of FIG. 1 will be described with reference to FIGS. This other specific example includes a decoder 58 shown in FIG. 9 and a signal switching unit 65 shown in FIG. The signal processing device 5 having the decoder 58 and the signal switching unit 65 also reads out the voice parameter codes shown in Tables 1 and 2 recorded in the memory 13 in FIG. Can be switched and output as desired by the user.

The encoding method in the speech encoder in the signal processing device 21 of the transmitting device 15 shown in FIG.
If it is determined to be based on the P (Vector Sum Excited Linear Prediction) coding method, the decoder 58 decodes the sound using the transmission signal by the VSELP coding, and outputs the decoded sound Snd _N to the output terminal 59. , The linear prediction coefficient α _N at the output terminal 60, the excitation source 1Exc _N1 at the output terminal 61, and the excitation source 2 at the output terminal 62.
Supply Exc _N2 .

The signal switching section 65 has a configuration as shown in FIG. The difference from the signal switching unit 32 shown in FIG. 4 is that an excitation source switching & extension unit 68 is provided.

[0093] The PSI-CELP performs a process for allowing the codec itself, particularly the voiced sound V, to be heard audibly smoothly. However, the VSELP does not have this, and therefore, when the bandwidth is expanded, some noise is mixed. Sounds like.
Therefore, when creating a broadband excitation source, an excitation source switching & extension unit 68 having a unit for switching the excitation source is used.
The processing as shown in FIG. 11 is performed. The processing shown in FIG. 11 is obtained by changing the excitation source processing shown in FIG. 6 to steps S87 to S89.

The excitation source of VSELP is represented by β * using parameters β (long-term prediction coefficient), bL [i] (long-term filter state), γ (gain), and c1 [i] (excitation code vector) used for the codec. bL [i] + γ * c1 [i], the former of which represents the pitch component and the latter of which represents the noise component, which are represented by β * bL [i] and γ * c1
[i], and if the former energy is large in a certain time range in step S87, the voice is considered to be a voiced sound having a strong pitch. Therefore, the process proceeds to YES in step S88, the excitation source is set to a pulse train, and the pitch component When there was no part, the process proceeded to NO and suppressed to zero. If the energy is not large in step S87, the conventional narrow band excitation source is filled up with zero by PZ-CELP in step S89 by zero padding in step S89 to obtain a wide band excitation source. As a result, the auditory quality of voiced sound in VSELP is improved.

Then, in step S92, the upsampling section 45 up-samples the original sound Snd _N and adds it in step S95 with the adding section 46. At this time, the high-frequency side is filtered by the high-frequency extraction and suppression filter 42 that slightly suppresses the component of about 6 KHz or more in step S91, so that the sound is easy to hear. This filter coefficient is selectable as described above.

Further, in step S93, the multiplication section 43
To adjust the high-frequency gain according to the user's preference.

As described above, the signal switching section 65 using the VSELP decoding method also uses the first band B ₁ (300 to
3400Hz) audio signal, sampling frequency is 16
Can be sent to the D / A converter 6 first audio signal or the sampling frequency of the band B ₁ KHz is switched speech signal of the wide band B _W of 16 KHz.

The voice parameter code recorded in the memory 13 at the time of answering machine recording is read out at the time of answering machine playback, and
The above three types of audio signals can be generated, switched and output. In particular, the sampling frequency is 8KHz to 16KH
High quality wideband speech converted to z can be generated.

For this reason, the receiver 1 including the signal switching unit 65 as the signal processor 5 has the first band B ₁ (3) whose sampling frequency is different from 8 KHz and 16 KHz.
(00 Hz to 3400 Hz), a received voice signal by VSELP, and a wide band (3
A received voice signal by VSELP (00 Hz to 6000 Hz) can be reproduced according to the user's selection at the time of real-time reproduction and the answering machine green reproduction. The user has more options. In addition to expanding the band of the received voice according to the situation, the received voice can be made the same as the band at the time of input, so that the built-in battery can be reduced.

Also, a voice parameter code having a sampling frequency of 8 KHz is recorded in the memory 13, and a wideband voice having a sampling frequency of 16 KHz can be used as reproduced voice. Audio can be played.

Note that the sampling frequency in the D / A converter 6 may be fixed at 16 KHz. Further, in the up-sampling unit 45, when the changeover switch 47 switches the sampling frequency between 8KHz / 16KHz,
Clear the filter output.

Further, as the signal processing device 5 inside the receiving device 1 in FIG. 1, a signal processing device comprising a signal switching unit 70 shown in FIG. 12 and a decoding unit shown in FIG. It may be. The signal processing device 5 including the decoder unit and the signal switching unit 70 also performs the operation shown in FIG.
The voice parameter codes shown in Tables 1 and 2 recorded in the memory 13 can be read out from the memory during answering machine playback, and the three types of voice signals can be switched and output as desired by the user.

The decoding unit shown in FIG.
A decoder 77 and a PSI-CELP decoder 81 are provided, and input of a speech parameter code to the decoder 77 or 81 according to the transmission signal encoding method transmitted from the transmitting device or recorded in the memory. Switch. That is, the transmission signal received via the input terminal 75 is switched by the changeover switch 76 in accordance with the type of the encoding method, that is, VSELP or PSI-CELP.

The two excitation sources 1Exc _N1 and 2Exc _N2 from the VSELP decoder 77 are supplied to the input terminals 66 and 67 of FIG. The excitation source NExc _N from the PSI-CELP decoder 81 is connected to the input terminal 3 of FIG.
5 is supplied.

The VSELP decoder 77 or PSI
-Linear prediction coefficient α _V or α _p from CELP decoder 81
Is selected by the changeover switch 80 in accordance with the type of the encoding system, and is supplied to the input terminal 34 of FIG.

Similarly, the VSELP decoder 77 or PS
The decoded sound from the I-CELP decoder 81 is also selected by the changeover switch 84 in accordance with the type of the above-mentioned encoding system, and then supplied to the input terminal 33 of FIG.

On the signal switching unit 70 side shown in FIG. 12, a switching switch 71 that switches according to the type of the above-mentioned encoding system causes an excitation source switching & extension unit 68 or an excitation source from the excitation source extension unit 41 to be switched. The output is switched and supplied to the LPC synthesis unit 40.

Therefore, also in this signal switching section 70, the sampling frequency is set to 8 KHz based on the user's preference.
First band B ₁ audio signal (ranging from 300 to 3400 Hz) of the first band of the sampling frequency is 16 KHz B ₁
Of the audio signal or the sampling frequency can be sent to the D / A converter 6 by switching the audio signal of the wide band B _W of 16 KHz.

Also, the voice parameter code recorded in the memory 13 at the time of answering machine recording is read out at the time of answering machine playback, and
The above three types of audio signals can be generated, switched and output. In particular, the sampling frequency is 8KHz to 16KH
High quality wideband speech converted to z can be generated.

For this reason, the receiver 1 including the signal switching unit 70 as the signal processor 5 has the first band B ₁ (3) whose sampling frequency is different from 8 KHz and 16 KHz.
00Hz-3400Hz) VSELP or PSI-C
The received voice signal by ELP or the sampling frequency is 1
6KHz wideband (300Hz-6000Hz) VS
The received voice signal by ELP or PSI-CELP can be reproduced according to the user's selection at the time of real-time reproduction and the answering machine green reproduction. The user has more options. In addition to expanding the band of the received voice according to the situation, the received voice can be made the same as the band at the time of input, so that the built-in battery can be reduced.

Further, high-quality sound can be reproduced while the recording capacity of the memory 13 is reduced. In addition, D /
The sampling frequency in the A converter 6 may be fixed at 16 KHz. In the up-sampling unit 45, the sampling frequency of 8KH
The filter output is cleared before switching to z / 16 KHz.

Further, the signal processing device 5 inside the receiving device 1 shown in FIG. 1 may include a signal switching unit 90 as shown in FIG. The signal processing apparatus 5 having the band extending unit 90 also reads out the voice parameter codes shown in Tables 1 and 2 recorded in the memory 13 in FIG. Can be played.

An input terminal 91 of the signal switching section 90 is supplied with an excitation source which is an LPC residual read from the memory 13 at the time of real-time operation or at the time of answering machine reproduction. Similarly, the input terminal 92 is supplied with the linear prediction coefficient α read from the memory 13 at the time of real-time operation or at the time of answering machine reproduction. The excitation source from the input terminal 91 is sent to the LPC synthesis filter 93 and also to the up-sampling unit 100. The linear prediction coefficient from the input terminal 92 is sent to the LPC synthesis filter 93.

The LPC synthesis filter 93 has an input terminal 91
The speech signal is synthesized using the linear prediction coefficient from the input terminal 92 on the basis of the excitation source from. LPC synthesis filter 93
Is supplied to the up-sampling unit 94, and the selected terminal a of the changeover switch 109 is
Supplied to

The up-sampling section 94 outputs the sampling frequency f of the audio signal synthesized by the LPC synthesis filter 93.
_{Upsample s1} . The upsampled audio signal is passed through only a predetermined band by a bandpass filter 95 and supplied to an adder 96. The addition output from the addition unit 96 is supplied to the selected terminal b of the changeover switch 109. The path leading to the up-sampling unit 94, the band-pass filter 95, and the adding unit 96 is a path for adding the signal of the component of the original frequency band to the synthesized audio signal.

The LPC synthesis filter 93 sends the linear prediction coefficient to the linear prediction coefficient / autocorrelation conversion section 97. The linear prediction coefficient-autocorrelation converter 97 converts the linear prediction coefficient into autocorrelation. This autocorrelation is sent to the narrowband codebook 98 and also to the affricate detector 99.

The excitation source from the input terminal 91 is up-sampled by the up-sampling unit 100 and sent to the LPC synthesis filter 103 via the low-pass filter 101 and the boost unit 102. The boost unit 102 boosts the excitation source when an affricate or a fricative is detected, and the boost amount of the boost unit 102 is controlled by an output of the affricate detector 99.

In the narrow band codebook 98, autocorrelation information of narrow band audio signals obtained from a plurality of audio signal patterns is stored in advance as code vectors. In the narrowband codebook 98, the linear prediction coefficient-autocorrelation conversion unit 9
7 and the autocorrelation information stored in the narrowband codebook 98, and a matching process is performed. Then, the index of the best matching autocorrelation information is sent to wideband codebook 104.

In the wideband codebook 104, corresponding to the narrowband codebook 98, autocorrelation information of a wideband audio signal obtained from an audio signal of the same pattern as when the narrowband codebook 98 was created is used as a code vector. Is stored. When the best matching autocorrelation information is determined in the narrowband codebook 98, this index is sent to the wideband codebook 104, and the wideband codebook 104 determines the narrowband autocorrelation information determined to be the best match. Broadband autocorrelation information corresponding to the correlation information is read.

The wideband autocorrelation information read from wideband codebook 104 is sent to autocorrelation / linear prediction coefficient conversion section 105. The autocorrelation-to-linear prediction coefficient conversion unit 105 converts the autocorrelation to a linear prediction coefficient. The LPC synthesis filter 103
Sent to

The LPC synthesis filter 103 performs LPC synthesis, and thereby synthesizes a wideband audio signal.
The audio signal synthesized by the LPC synthesis filter 103 is supplied to a high-frequency extraction and suppression filter 106 and a multiplier 107.

The high band extraction & suppression filter 106 is an LPC
A signal component in the frequency band of 300 Hz to 3400 Hz of the input narrow band audio signal is removed from the combined output from the combining filter 103 to extract a signal component of 3400 Hz or more, and suppresses a high frequency component according to the user's preference.
The multiplier 107 multiplies the filter output from the high-frequency extraction & suppression filter 106 by the gain adjusted from the terminal 108.

Then, the adder 96 adds the original narrowband audio signal component via the BPF 95 to the multiplied output from the multiplier 107. Thereby, a wideband audio signal is obtained. This audio signal is supplied to the selected terminal c of the changeover switch 109.

The changeover switch 109 controls the first band B of the first sampling frequency f _s1 (= 8 KHz) sent from the LPC synthesis filter 93 to the selected terminal a.
₁ (300 Hz to 3400 Hz) audio signal and audio of the first band B ₁ (300 Hz to 3400 Hz) of the second sampling frequency f _s2 (= 16 KHz) sent from the up-sampling unit 94 to the selected terminal b. The signal and the wide band B _w (300) of the second sampling frequency f _s2 (= 16 KHz) sent from the adding section 96 to the selected terminal c.
Any one of the audio signals is supplied to the D / A converter 6 by switching the audio signal with the audio signal of (Hz to 6000 Hz) based on the switching control signal from the switching control signal terminal 129 by the selection piece d.

As described above, even in the receiving apparatus provided with the signal switching section 90 shown in FIG. 14, the sampling frequency is set to 8 kHz by using the voice parameter code recorded in the memory 13 at the time of answering machine recording as well as at the time of real time operation. , 16 KHz and a first band B ₁ (30
0 Hz to 3400 Hz) or a wide band (300 Hz to 6000) with a sampling frequency of 16 KHz.
Hz) can be switched and output as desired by the user.

A receiving device using a signal processing device provided with the above signal switching units 32, 65, 70 or 90 is integrated with a transmitting device to constitute a portable telephone device 110 as shown in FIG. You may. This mobile phone device 110
It has an answering machine recording function. The voice parameter code is recorded in the semiconductor memory 130, read out in response to a user operation, and can be selected as one of the three types of voice signals.

In this portable telephone device 110, the audio signal input from the microphone 111 is stored in the memory 116a of the signal processing device 116 via the amplifier 112, the volume 113, the anti-aliasing filter 114 and the A / D converter 115. Is done.

The audio signal stored in the memory 116a is
The audio signal is encoded by an audio encoding unit in the signal processing device 116 and output as an audio parameter code.

This voice parameter code is transmitted to the control unit 117.
The signal is transmitted to the base station via the RF transceiver 118 and the antenna 119.

Here, the audio encoding unit in the signal processing device 116 supplies an audio parameter code to the RF transmission / reception 118 via the control unit 117 in consideration of the band narrowing limited by the transmission path.

Also, the speech parameter code received from the base station via antenna 119 is transmitted to RF transmitting / receiving section 118,
The memory 1 of the signal processing device 122 via the control unit 117
22a. When the answering machine recording function is set by the user, the voice parameter code is stored in the memory 130.

The voice parameter code stored in the memory 122a of the signal processing device 122 is decoded by the decoding section of the signal processing device 122, and then subjected to predetermined signal processing and output. Further, when the transmitted voice recorded by the answering machine is reproduced by the user's operation, the voice parameter code stored in the memory 130 is decoded by the decoding unit of the signal processing device 122 and then subjected to predetermined signal processing. Is output.

At the time of real-time operation and when the answering machine is regenerated, the output signal from the signal processing device 122 is output to the D / A converter 123.
Is converted into an analog signal, and is output from the speaker 127 via the anti-aliasing filter 124, the volume 125, and the amplifier 128.

Here, the signal processing device 122 includes the above-described signal switching unit 32, 65, 70 or 90. Therefore, the portable telephone device 110 stores the data in the memory 1 when recording the answering machine.
Using the speech parameter codes recorded in 30 when recorded message playback, the sampling frequency is 8 KHz, 16 KHz and different, and the reception voice signal in the first band B ₁ (300Hz~3400Hz), wideband sampling frequency of 16 KHz (300 Hz (6000 Hz) can be reproduced according to the user's selection.

Further, the voice parameter code having a sampling frequency of 8 KHz is recorded in the memory 130, and the reproduced voice is a wideband voice having a sampling frequency of 16KHz. Therefore, the recording is performed with a small recording capacity and the high quality voice is reproduced. can do.

In the receiving device and the portable telephone device, a semiconductor memory is used as a recording medium for recording the voice parameter code, but another recording medium such as a magnetic tape may be used.

Further, in the receiving device and the portable telephone device, the voice parameter code is recorded in the dedicated memory 13 and the memory 130. However, in the present invention, the memory in the control unit or the signal processing device may be shared.

[0138]

According to the present invention, the sampling frequency is different with the speech parameter codes recorded on the recording medium during voicemail playback, the audio signal and the first band B _1, the sampling frequency is a broadband audio signal It can be switched and output as desired by the user.

[0139] Further, since the audio parameter code having the sampling frequency _fs1 is recorded on the recording medium, the recording capacity of the recording medium can be saved.

That is, not only at the time of real-time operation, but also at the time of voice mail playback, it is possible to obtain a received voice with improved auditory quality as desired by the user.

[Brief description of the drawings]

FIG. 1 is a block diagram illustrating a configuration of a receiving device according to an embodiment of the present invention.

FIG. 2 is a block diagram showing a configuration of a transmitting apparatus for transmitting a voice parameter code to the receiving apparatus shown in FIG. 1 via a base station.

FIG. 3 is a diagram showing a PSI-CELP decoder which constitutes a signal processing device inside the receiving device shown in FIG. 1 together with a signal switching unit.

FIG. 4 is a functional block diagram for explaining a process of a signal switching unit configuring the signal processing device inside the receiving device shown in FIG. 1 together with a PSI-CELP decoder.

FIG. 5 is a block diagram showing a detailed configuration of an excitation source extension unit included in the signal switching unit shown in FIG. 4;

FIG. 6 is a flowchart for explaining a detailed operation of the signal switching unit shown in FIG. 4;

FIG. 7 is a flowchart for explaining a training data generation process used for a codebook used in the signal switching unit shown in FIG. 4;

FIG. 8 is a flowchart for explaining the generation of the code book.

FIG. 9 is a diagram illustrating a VSELP decoder included in another specific example of the signal processing device inside the receiving device illustrated in FIG. 1;

FIG. 10 is a functional block diagram for explaining processing of a signal switching unit included in another specific example of the signal processing device inside the receiving device shown in FIG. 1;

FIG. 11 is a flowchart illustrating a detailed operation of the signal switching unit shown in FIG. 10;

FIG. 12 is a functional block diagram for explaining processing of a signal switching unit included in still another specific example of the signal processing device inside the receiving device shown in FIG. 1;

FIG. 13 is a block diagram showing a configuration of a decoding unit included in still another specific example of the signal processing device inside the receiving device shown in FIG. 1;

FIG. 14 is a block diagram for explaining a process of a signal switching unit included in still another specific example of the signal processing device in the receiving device shown in FIG. 1;

FIG. 15 is a block diagram of a mobile phone device including a receiving device including a signal processing device using the signal switching unit shown in FIG. 4, FIG. 10, FIG. 12, or FIG. FIG. 3 is a block diagram illustrating a configuration.

[Explanation of symbols]

1 receiver, 5 signal processor, 13 memory, 15
Transmitter, 21 signal processor, 27 PSI-CE
LP decoder, 32 signal switching unit, 36 linear prediction coefficient → autocorrelation (α _N → r _N ) conversion unit, 37 autocorrelation broadband unit, 38 wideband codebook, 39 autocorrelation → linear prediction coefficient conversion unit, 40 LPC synthesis Section, 41 excitation source extension section, 47 changeover switch

 ────────────────────────────────────────────────── ─── Continued on the front page (72) Inventor Shiro Omori 6-7-35 Kita-Shinagawa, Shinagawa-ku, Tokyo Sony Corporation F-term (reference) 5D045 CA01 CB01 5K052 AA00 BB02 BB11 CC06 EE38 EE40 FF05 GG48 9A001 BB02 BZ03 EE02 EE05 GG05 GG07 HH18 JJ12

Claims (19)

[Claims] 1. A recording medium for recording an audio parameter code based on a transmission signal transmitted from a transmission device to generate an audio signal of a first sampling frequency f _s1 , and the audio recorded on the recording medium the sampling frequency of the first band B ₁ of the audio signal generated using the parameter codes second sampling frequency _{f s2 (f s2> f s1} )
A sampling rate converting means for converting the second sampling frequency f _s2 of the recording medium with the recorded the voice parameter codes in the second band B ₂ is a band component of the first band B ₁ Out-of-band component estimating means for estimating the audio signal of the above, an audio signal having the second sampling frequency f _{s2 set} by the sampling rate converting means, and a second signal estimated by the out-of-band component estimating means on the audio signal. receiving apparatus characterized by comprising a switching means for switching between audio signals of the second wide band B _W of the audio signal of the band B ₂ obtained by adding the sampling frequency f _s2. 2. The switching means also switches an audio signal of a _first band _B1 of a first sampling frequency _fs1 generated using the audio parameter code recorded on the recording medium. The receiving device according to claim 1, wherein 3. The out-of-band component estimating means includes a part for band-extending the linear prediction residual as the speech parameter code and a part for extending the linear prediction coefficient as the speech parameter code to a wide band. Claim 1.
The receiving device according to the above. 4. The extended part of the linear prediction coefficient to a wide band is a first conversion unit that converts the linear prediction coefficient into an autocorrelation, and a code in which the autocorrelation of the first conversion unit stores a wideband autocorrelation in advance. 4. The apparatus according to claim 3, further comprising: an autocorrelation expansion unit that expands by referring to the book; and a second conversion unit that converts the expanded autocorrelation from the autocorrelation expansion unit into expanded linear prediction coefficients. Receiver. 5. The receiving apparatus according to claim 3, wherein the portion for band-extending the linear prediction residual includes an up-sampling unit for up-sampling the linear prediction residual. 6. The transmission signal is a PSI-CELP coded or VSELP coded signal, and the out-of-band component estimating means is configured to perform the PSI-CELP coded or VSEL
P coded signal using a speech parameter codes in the obtained the recording medium by decoding a band component of the first band B ₁ second second sampling frequency f of the band B ₂ The receiving device according to claim 1, wherein the audio signal of _s2 is estimated. 7. The sampling rate conversion means, when the switching means switches from the first sampling frequency f _s1 to the second sampling frequency f _s2 , initializes a filter state. The receiving device according to claim 1. 8. A sampling of the first first band B ₁ of the audio signal generated by reading from the speech parameter codes the recording medium based on the transmission signal transmitted to produce the audio signal having the sampling frequency f _s1 A converted output obtained by converting the frequency to a second sampling frequency f _s2 (f _s2 > f _s1 ), and a first band B estimated using the audio parameter code read from the recording medium for the converted output. reception method characterized by switching between the sum output plus second audio signal of a second sampling frequency f _s2 of band B ₂ is out-of-band component of the _1. 9. The reception according to claim 8, wherein an audio signal of a _first band _B1 of a first sampling frequency _fs1 generated using an audio parameter code read from the recording medium is also switched. Method. 10. A transmission means for performing an encoding process on a first sampling frequency f _{s1 on} an input audio signal to generate a transmission signal, and a transmission unit generating the transmission signal on the basis of the first sampling frequency f _s1 . first sampling frequency of the audio signal band B1 second sampling frequency f _s2 converted output and obtained by converting the (f _s2> f _s1) generated by reading from the speech parameter codes the recording medium based on the transmission signal A second sampling frequency f estimated using the audio parameter code read from the recording medium for the converted output.
communication device comprising: a receiving means for switching between addition output obtained by adding the second audio signal having a bandwidth B ₂ of _s2. 11. A recording medium for recording an audio parameter code based on a transmission signal transmitted from a transmission device to generate an audio signal of the first sampling frequency f _s1 , wherein: a sampling rate converting means for converting the code into the recording first sampling frequency of the band B ₁ of the audio signal a second sampling frequency f generated is read from the medium _{s2 (f s2> f s 1} ), said recording medium and out-of-band component predicting unit for using the speech parameter codes to infer an audio signal of the second sampling frequency f _s2 of the second band B ₂ is a band component of the first band B ₁ read from, and the audio signal with the second sampling frequency f _s2 at the sampling rate converting means, was estimated by the band components estimating means to the audio signal Communication device according to claim 10, characterized in that it comprises a switching means for switching between audio signals of the second wide band B _W of the audio signal of the band B ₂ obtained by adding the two's sampling frequency f _s2. 12. The apparatus according to claim 11, wherein said switching means also switches an audio signal of a _first band _B1 of a first sampling frequency f _s1 generated using said audio parameter code. Communication device. 13. The out-of-band component estimating means of the receiving means includes a part for extending a band of the linear prediction residual as the speech parameter code, and a part for extending a linear prediction coefficient as the speech parameter code to a wide band. The communication device according to claim 11, comprising: 14. An extended part of the linear prediction coefficient to a wide band is a first conversion unit for converting the linear prediction coefficient into an autocorrelation, and a code in which the autocorrelation of the first conversion unit stores a wideband autocorrelation in advance. 14. The autocorrelation expansion unit that expands by referring to a book, and a second conversion unit that converts the expanded autocorrelation from the autocorrelation expansion unit into expanded linear prediction coefficients. Communication device. 15. The communication apparatus according to claim 13, wherein the portion that extends the band of the linear prediction residual includes an upsampler that upsamples the linear prediction residual. 16. The transmission signal is a PSI-CELP coded or VSELP coded signal, and the out-of-band component estimating means performs the PSI-CELP coded or VSELP coded signal.
The voice parameter code obtained by decoding the LP-encoded signal is read out from the recording medium, and the first band B
Communication device according to claim 11, characterized in that inferring _a second audio signal of a second sampling frequency f _s2 of band B ₂ is out-of-band component of the. 17. The sampling rate conversion means,
_12. The filter state is initialized when the switching means switches from the first sampling frequency f _s1 to the second sampling frequency f _s2.
The communication device as described. 18. and generates a transmission signal by performing encoding processing by the first sampling frequency f _s1 for the input speech signal, has been transmitted to produce a speech signal of the first sampling frequency f _s1 the speech parameter codes the second sampling frequency of the band B ₂ is out-of-band component of the band B ₁ was the first product is read from the recording medium based on a transmission signal a second sampling frequency f _s2 (f _s2
> The conversion output obtained by converting the f _s1), and the converted output to the addition obtained by adding the second audio signal having a bandwidth B ₂ of the second sampling frequency f _s2 of guessed using the speech parameter codes outputted A communication method characterized by switching between: 19. The communication method according to claim 18, wherein an audio signal of a _first band _B1 of a first sampling frequency _fs1 generated using an audio parameter code read from the recording medium is also switched. .