JP2000206997A - 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 by which hearing quality of the reproducing of an automatic recording is improved. 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 and generates wide band voices having a sampling frequency of 16 kHz. Thus, in a receiving device 1, the recording capacity of the memory 13 is saved and yet voices having a high quality are reproduced.

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, at the time of the above-mentioned answering machine reproduction, the sound quality of the sound output in the same sound frequency band as the input sound, whose transmission band is limited, cannot be said to be very good. That is, the auditory quality is inferior. They also complain about the sound quality of digital mobile phones.

The present invention has been made in view of the above circumstances, and has as its object to provide a receiving apparatus and method, a communication apparatus, and a method capable of obtaining a received voice with improved auditory quality during answering machine reproduction. I do.

[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.
Out-of-band component estimating means for estimating an audio signal of the second sampling frequency f _s2 of the band B ₂ of the above.

[0007] Then, the receiving apparatus, and the first audio signal band B _1, which is the second sampling frequency f _s2 at the sampling rate converting means, the was estimated by the out-of-band components estimating means the 2 sampling frequency f
_s2 of the said second sound signal in band B ₂ of adding with a summing means.

According to another aspect of the present invention, there is provided a receiving method, comprising the steps of: storing a voice parameter code based on a transmission signal transmitted to generate a voice signal of a first sampling frequency f _{s1 on} a recording medium; read from to generate a first audio signal having a bandwidth B ₁ in, it converts the sampling frequency to a second sampling frequency _{f s2 (f s2> f s1} ), and the speech parameter codes read from the recording medium use and guess 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.

In order to solve the above-mentioned problems, a communication apparatus according to the present invention performs a coding process at an input audio signal at a first sampling frequency f _s1 to generate a transmission signal; Receiving means for reading an audio parameter code based on a transmission signal generated by performing an encoding process at a sampling frequency f _s1 from a recording medium to generate an audio signal having a second sampling frequency f _s2 (f _s2 > f _s1 ); Is provided.

Here, the receiving means includes: a recording medium for recording an audio parameter code based on a transmission signal transmitted from a transmission 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 Out-of-band component estimating means for estimating an audio signal of a second sampling frequency f _s2 of a second band B ₂ which is an out-of-band component of the first band B ₁ using the audio parameter code read from Prepare.

[0011] Then, the receiving means, and said first audio signal band B _1, which is the second sampling frequency f _s2 at the sampling rate converting means, said first was estimated by the out-of-band components estimating means 2 sampling frequency f
_s2 of the said second sound signal in band B ₂ of adding with a summing means.

Further, in order to solve the above problem, the 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, An audio parameter code based on a transmission signal generated by performing an encoding process at a frequency f _{s1 is} read from a recording medium, and a second sampling of a second band B ₂ which is an out-of-band component of the first band B ₁ is performed. Estimate the audio signal of frequency f _s2 .

In the present invention, the first sampling frequency f _s1 of the first band B ₁ generated by reading out the audio parameter code based on the transmission signal sent to the receiving means from the recording medium is used as the second sampling frequency. Frequency f _s2 ( _fs2
> F _s1 ) and the second band B ₂ of the second sampling frequency f _s2 estimated using the above speech parameter code.
, A wideband sound having a high sampling frequency can be obtained.

[0014]

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 obtains a second sampling frequency f _s2 (from a voice parameter code transmitted from a transmitting apparatus to be described later via a base station in order to generate a voice signal of a first sampling frequency f _s1 (f _s1 ). _fs2 > _fs1 ) is generated. 8 KHz is used as the first sampling frequency f _{s1 and} 16 KHz is used as the second sampling frequency f _s2 .

The receiving apparatus 1 has a so-called answering machine recording function. The voice parameter code is recorded in, for example, a semiconductor memory, and the voice parameter code is stored in the memory in accordance with a user operation. The transmitted voice transmitted during reading and answering can be reproduced at the second sampling frequency _fs2 .

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 receiving unit 3 and the control unit 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, and 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, an output signal from the signal processing device 5 is converted into an analog signal by the D / A converter 6 and then passed through the anti-aliasing filter 7, the volume 8 and the 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 speech coding unit in the signal processing device 21 generates a speech parameter code in consideration of the narrow band limited by the radio 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 that in the digital telephone standard,
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.

[0026]

[Table 1]

[0027]

[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 pitch frequency as shown in Tables 1 and 2, For example, there are frame power R0, CRC, and the like in a frame of 20 msec. These speech parameter codes are transmitted in a 10-word configuration in which 16 bits (bit) are 1 word (Word) per frame.

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 bandwidth extension unit 32 shown in FIG.

The encoding method in the speech encoder 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 bandwidth extension unit 32 uses the PSI-CELP encoded transmission signal transmitted from the transmission device to generate the audio signal of the first sampling frequency f _s1 (= 8 KHz). Decoded sound Snd _N of decoded _first band B ₁ (300 to 3400 Hz)
Of the second sampling frequency f _s2
(= 16 KHz), and a linear prediction coefficient α obtained by the decoder 27 decoding the transmission signal by the PSI-CELP encoding.
_N and the excitation source NExc _N , the second band B ₂ (3400) of the second sampling frequency f _s2 (= 16 KHz).
(Hz to 6000 Hz).

Then, the first sampling frequency _fs2 is set to the second sampling frequency _fs2 by the sampling rate conversion means.
Audio S in the band B ₁ (300-3400 Hz)
nd _N and a signal in the second band B2 of the _second sampling frequency f _{s2 which is} an out-of-band component of the first band B ₁ (300 to 3400 Hz) estimated by the out-of-band component estimating means. Are added by an adder 46 serving as an adding means.

Here, the sampling rate conversion means is the up-sampling section 45 in FIG. The out-of-band component estimating means is a part excluding the up-sampling unit 45 and the adder 46.

Hereinafter, the configuration of the bandwidth extension unit 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 multiplier 43.

The linear prediction coefficient α _N supplied from the input terminal 34 is converted by a 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.

[0037] 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 multiplier 43 by a gain supplied from a terminal 44. 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 via 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.

Further, as described above, the bandwidth extension section 32 serves as the sampling rate conversion means as the input terminal 3.
3; first band B ₁ = 300-3400
Hz of the sampling frequency of the decoded audio Snd _N
An up-sampling unit 45 for up-sampling from _s1 = 8 kHz to _fs2 = 16 kHz is provided.

The sampling frequency of the up-sampling section 45 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. As described above, the bandwidth extending unit 32 outputs the wideband audio signal Snd _w having the band of 300 to 6000 Hz and the sampling frequency of 16 kHz from the output terminal 47.

In the bandwidth extension unit 32 having the above configuration,
The principle of operation will be described below. Bandwidth extension unit 32, from the speech parameter codes to generate a first audio signal having a bandwidth B ₁ in 300Hz～3400Hz 34
A speech coding parameter for a _second band B2 of 00 Hz to 6000 Hz is generated, and wideband LPC synthesis is performed.
After that, the low frequency band (300 Hz to
3400 Hz) side is replaced with the original sound upsampled to 16 KHz. That is, a high-pass filter is applied to keep only the high frequency band (3400 Hz to 6000 Hz), suppress high frequency components among the high frequency components, further adjust the gain, and then adjust the original sound (300 Hz to 3 000 Hz).
400 Hz) is added to the up-sampled (second sampling frequency f _s2 ).

Here, to widen (or extend) 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 is established between the narrowband autocorrelation and the wideband autocorrelation as described later, only a codebook based on the wideband autocorrelation needs to 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).

[0048]

(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.

[0051]

(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) becomes the following equation (3)
It looks like an expression.

[0053]

(Equation 3)

The pass band and the stop band of this new filter are equal to 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.

[0055]

(Equation 4)

As described above, when performing 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.

[0057] 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 it can be more accurately expanded by dividing it into voiced sound (V) and unvoiced sound (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 voice synthesized by the wideband LPC 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, a detailed operation of the bandwidth extension unit 32 based on the above-described operation principle will be described with reference to a flowchart of FIG.

In step S1, the α _N → r _N conversion unit 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 autocorrelation r _N for voiced sound 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, when the judgment result in step S2 is UV, in step S4 the autocorrelation r for unvoiced sound is quantized using the narrow band UV parameter 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 filling zeros between samples by the up-sampling unit 50 shown in FIG. 5 in step S7, and 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, in this case, the signal is merely a wideband signal obtained by prediction, and an error due to prediction is included, so that the quality is poor. 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 in step S13 by the adder 46. 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 multiplier 43
To adjust the high-frequency gain according to the user's preference.

Here, the creation of a codebook used in the bandwidth extension unit 32 will be described. The creation of the codebook is generally well-known GLA (Generalized Llo
yd Algorithm). The wideband speech is divided into frames for a certain period of time, for example, every 20 msec, and the autocorrelation of a certain order, for example, the sixth order is obtained for each frame. Using the autocorrelation for each frame as training data, a six-dimensional codebook is created. At this time, the voiced sound and the unvoiced sound may be distinguished from each other, and the autocorrelation of the voiced sound and the autocorrelation of the unvoiced sound 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 bandwidth extending section 32 uses a codebook for a wideband voiced sound and a codebook for a wideband unvoiced sound. The creation of the codebook for wideband voiced sounds will be described with reference to FIG. 7, and the creation of the codebook for wideband unvoiced sounds 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 in step S42 by GLA.

As described above, in the bandwidth extension unit 32 using the decoding method based on PSI-CELP, the sampling frequency is converted from 8 KHz to 16 KHz by using the voice parameter code recorded in the memory 13 at the time of answering machine recording and at the time of answering machine playback. High-quality wideband speech can be generated.

The memory 13 has a sampling frequency of 8K.
Since the audio parameter code of Hz is recorded and the reproduced audio is a wideband audio having a sampling frequency of 16 KHz, it is possible to record with a small recording capacity and reproduce high quality audio.

Next, another specific example of the signal processing device 5 inside the receiving device 1 of FIG. 1 will be described with reference to FIGS. Another specific example includes a decoder 58 shown in FIG. 9 and a bandwidth extension unit 65 shown in FIG. The signal processing apparatus 5 including the decoder 58 and the bandwidth extending unit 65 also reads out the voice parameter codes shown in Tables 1 and 2 recorded in the memory 13 in FIG. Can reproduce 16KHx wideband audio.

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

[0086] 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, noise is mixed slightly. 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 up-sampler 45 up-samples the original sound Snd _N and adds it in step S95 with the adder 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 multiplier 43
To adjust the high-frequency gain according to the user's preference.

As described above, the bandwidth extension unit 65 using the VSELP decoding method also uses the voice parameter code recorded in the memory 13 at the time of answering machine recording to convert the sampling frequency from 8 KHz to 16 KHz at the time of answering phone recording playback. High quality wideband speech can be generated.

The memory 13 has a sampling frequency of 8K.
Since the audio parameter code of Hz is recorded and the reproduced audio is a wideband audio having a sampling frequency of 16 KHz, it is possible to record with a small recording capacity and reproduce high quality audio.

Further, as the signal processing device 5 inside the receiving device 1 shown in FIG. 1, a signal processing device comprising a bandwidth extending unit 70 shown in FIG. 12 and a decoding unit shown in FIG. An example may be used. The signal processing apparatus 5 having the decoder unit and the bandwidth extending unit 70 also reads out the voice parameter codes shown in Tables 1 and 2 recorded in the memory 13 in FIG. Can reproduce 16KHx wideband audio.

The decoding unit shown in FIG.
It includes a decoder 77 and a PSI-CELP decoder 81, and switches the input of a speech parameter code to the decoder 77 or 81 in accordance with the encoding method of the transmission signal transmitted from the transmitting device. That is, the transmission signal received via the input terminal 75 is changed by the changeover switch 76 to the type of the encoding method, that is, VSELP or PSI-C.
Switching is performed according to the ELP.

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.

Further, on the side of the bandwidth extension unit 70 shown in FIG. 12, the excitation switch from the excitation source switching & extension unit 68 or the excitation source extension unit 41 is switched by the changeover switch 71 which switches according to the type of the above-mentioned encoding method. The source output is switched and supplied to the LPC synthesis unit 40.

Therefore, in this bandwidth extension unit 70,
The voice parameter code recorded in the memory 13 at the time of answering machine recording is used at the time of answering machine playback, and the sampling frequency is 8K.
It is possible to generate high-quality wideband sound converted from Hz to 16 kHz. In particular, if the sampling frequency is 2
The doubled high-quality bandwidth extension can be performed using the voice parameter code recorded in the memory 13 in accordance with the type of the encoding method of the transmission signal transmitted from the transmission device side.

Further, the signal processing device 5 inside the receiving device 1 of FIG. 1 may include a bandwidth extending unit 90 as shown in FIG. The signal processing device 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.
x wideband audio can be reproduced.

An excitation source which is an LPC residual read out from the memory 13 is supplied to an input terminal 91 of the bandwidth extension unit 90. The input terminal 92 is supplied with the linear prediction coefficient α read from the memory 13. The excitation source from the input terminal 91 is sent to the LPC synthesis filter 93 and also to the up-sampling unit 100. Input terminal 92
Are 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.

The up-sampling section 94 provides a sampling frequency f of the audio signal synthesized by the LPC synthesis filter 93.
_{Upsample s1} . The upsampled audio signal passes through only a predetermined band by a bandpass filter 95 and is supplied to an adder 96. The path leading to the up-sampling section 94, the band-pass filter 95, and the adding section 96 is a path for adding the signal of the component of the original frequency band to the synthesized audio signal.

The linear prediction coefficient is sent from the LPC synthesis filter 93 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, the autocorrelation information of the wideband audio signal obtained from the audio signal of the same pattern as when the narrowband codebook 98 was created is stored as a code vector in correspondence with the narrowband codebook 98. 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, thereby synthesizing 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.

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 output from the output terminal 109.

As described above, even in the receiving apparatus including the bandwidth extending section 90 shown in FIG.
The voice parameter code recorded in No. 3 can be used at the time of answering machine reproduction to generate a high-quality wideband voice whose sampling frequency has been converted from 8 KHz to 16 KHz.

The bandwidth extension units 32, 65, 70
Alternatively, the receiving device using the signal processing device provided with 90 may be integrated with the transmitting device to constitute a mobile phone device 110 as shown in FIG. This mobile phone device 110
Has an answering machine recording function, records the voice parameter code on a recording medium such as a semiconductor memory or a magnetic tape, and reads out the voice parameter code from the recording medium in response to a user operation. The transmitted voice transmitted during the absence can be reproduced at the second sampling frequency _fs2 .

In the 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.

The 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 unit 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 the antenna 119 is
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 speech 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 the real-time operation and when the answering machine is regenerated, the output signal from the signal processing device 122 is supplied 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-mentioned bandwidth extending unit 32, 65, 70 or 90. Therefore, the mobile phone device 110 uses the voice parameter code recorded in the memory 130 at the time of answering machine recording and uses the voice frequency code at the time of answering machine playback to change the sampling frequency from 8 kHz to 16 kHz.
, And a high-quality wideband sound converted to the high-quality sound can be generated. The memory 130 has a sampling frequency of 8K.
Since the audio parameter code of Hz is recorded and the reproduced audio is a broadband audio having a sampling frequency of 16 KHz, it is possible to record with a small recording capacity and reproduce high quality audio.

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 other recording media such as a magnetic tape may be used.

Further, in the above-mentioned receiving device and portable telephone device, the above-mentioned 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.

[0123]

According to the present invention, it is possible to generate a high-quality wide-band sound whose sampling frequency is doubled, for example, by using the sound parameter code recorded on the recording medium at the time of answering machine reproduction.

[0124] Further, the sampling frequency on a recording medium to record the speech parameter codes of f _s1, since the sampling frequency as the playback sound is to wideband speech of _{f s2 (f s1 <f s2} ), the recording capacity of the recording medium Can be saved.

[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 configures a signal processing device inside the receiving device shown in FIG. 1 together with a bandwidth extension unit.

FIG. 4 is a functional block diagram for explaining processing of a bandwidth extension unit that configures 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 bandwidth extension unit shown in FIG. 4;

FIG. 6 is a flowchart illustrating a detailed operation of the bandwidth extension unit shown in FIG. 4;

FIG. 7 is a flowchart for explaining training data generation processing used for a codebook used in the bandwidth extension 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 bandwidth extension 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 bandwidth extension unit shown in FIG. 10;

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

FIG. 15 is a portable telephone device having a receiving device including a signal processing device using the bandwidth extending section shown in FIG. 4, FIG. 10, FIG. 12, or FIG. FIG. 3 is a block diagram showing the configuration of FIG.

[Explanation of symbols]

1 receiver, 5 signal processor, 13 memory, 15
Transmitter, 21 signal processor, 27 PSI-CE
LP decoder, 32 bandwidth extension unit, 36 linear prediction coefficient → autocorrelation (α _N → r _N ) conversion unit, 37 autocorrelation wideband unit, 38 wideband codebook, 39 autocorrelation → linear prediction coefficient conversion unit, 40 LPC Synthesis unit, 41 Excitation source extension unit

 ────────────────────────────────────────────────── ─── Continuing on the front page (72) Inventor Shiro Omori 6-7-35 Kita Shinagawa, Shinagawa-ku, Tokyo Sony Corporation F-term (reference) 5D045 CB01 5K052 AA00 BB02 BB12 EE40 FF07 GG34 GG48 9A001 CC02 EE05 JJ12 KK56

Claims (15)

[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 And a out-of-band component estimating means for estimating the audio signal. 2. The method according to claim 1, wherein said sampling rate conversion means includes a second
The first band B _{1 set} to the sampling frequency f _{s2 of}
And the second band B _{2 of} the second sampling frequency f _s2 estimated by the out-of-band component estimating means.
The receiving apparatus according to claim 1, further comprising an adding unit that adds the audio signal. 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. A voice parameter code based on a transmission signal transmitted to generate a voice signal of a first sampling frequency f _s1 is read from a recording medium to generate a voice signal of a _first band B ₁ , Converting the sampling frequency to a second sampling frequency f _s2 (f _s2 > f _s1 ),
And using the audio parameter code read from the recording medium, a second out-of-band component of the _first band B1.
And estimating an audio signal of the second sampling frequency f _s2 of the band B ₂ . 8. A transmission means for performing an encoding process at a first sampling frequency f _{s1 on} an input audio signal to generate a transmission signal; and a transmission unit for generating an encoding process at the first sampling frequency f _s1 . The voice parameter code based on the transmission signal is read out from the recording medium and the second sampling frequency f
receiving means for generating an audio signal of _s2 ( _fs2 > _fs1 ). 9. 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 Out-of-band component estimating means for estimating an audio signal of a second sampling frequency f _s2 of a second band B ₂ which is an out-of-band component of the first band B ₁ using the audio parameter code read from The communication device according to claim 8, further comprising: 10. The receiving means and said first audio signal band B _1, which is the second sampling frequency f _s2 at the sampling rate converting means, said first was estimated by the out-of-band components estimating means communication device according to claim 9, characterized in that it comprises an adding means for adding the 2 of the second audio signal having a bandwidth B ₂ of the sampling frequency f _s2. 11. The out-of-band component estimating means of the receiving means includes a part for extending the band of 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. 10. The communication device according to claim 9, comprising: 12. The wideband extension of the linear prediction coefficient is performed by 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. 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. 13. The communication apparatus according to claim 11, wherein the portion that extends the bandwidth of the linear prediction residual includes an upsampler that upsamples the linear prediction residual. 14. The transmission signal is a PSI-CELP coded or VSELP coded signal, and the out-of-band component estimating means includes a 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
₁ communication apparatus according to claim 9, wherein the inferring audio signal of the second sampling frequency f _s2 of the second band B ₂ is out-of-band component of the. 15. An input audio signal is subjected to encoding processing at a first sampling frequency f _s1 to generate a transmission signal,
An audio parameter code based on the transmission signal generated by performing the encoding process using the first sampling frequency f _{s1 is} read out from a recording medium, and a second band B ₂ which is an out-of-band component of the first band B ₁ is read. A communication signal characterized by estimating the audio signal of the second sampling frequency _fs2 .