CN106328153B - Electronic communication equipment voice signal processing system and method and electronic communication equipment - Google Patents
Electronic communication equipment voice signal processing system and method and electronic communication equipment Download PDFInfo
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Abstract
The invention provides a voice signal processing system of electronic communication equipment, which comprises a voice signal acquisition unit, a voice processing unit and a voice processing unit, wherein the voice signal acquisition unit is used for acquiring a voice signal of an information source and converting the voice signal into a setting signal; the voice signal processing unit is used for grouping the set signals according to the channel bandwidth lower limit threshold value, processing at least two groups of voice signals with different bandwidths in the set signals, generating harmonic signals of a group of voice signals with a lower bandwidth threshold value, and overlapping the harmonic signals with a group of voice signals with a higher bandwidth threshold value to generate secondary set signals; and a voice signal pickup unit for picking up the secondary setting signal and outputting it through a channel. A signal processing method and an electronic device having the signal processing system are also disclosed. The invention has the advantages of stable improvement of the quality of the voice signal and high restoration degree of the voice signal.
Description
Technical Field
The present invention relates to the field of communications technologies, and in particular, to a system and a method for processing a voice signal of an electronic communication device, and an electronic communication device.
Background
In the traditional communication field, the frequency range of the transmitted voice signal is 300Hz to 3400Hz, and the set bandwidth is about 3 KHz. In practice, the natural speech signal spectrum is 100Hz to 7000 Hz. Therefore, in actual communication, the channel will automatically filter out speech signals above or below its upper or lower frequency limit. When speech signals above or below their upper or lower frequency limit are filtered out, the speech signals in the communication channel will naturally be distorted to some extent.
In order to optimize the transmission effect of the voice signal, in the prior art, the signal at the receiving end is usually processed to compensate the distortion of the voice signal in the communication signal. For example, the device and method for band extension of a speech signal disclosed in patent publication No. 105408975 is directed to spread an input signal having harmonics in the input signal at a low bit rate to obtain a better sound quality.
Therefore, a method for compensating the voice signal at the signal transmitting end and further improving the call quality is lacked in the prior art.
Disclosure of Invention
The invention aims to provide a method for compensating a voice signal at a signal sending end to improve the call quality.
The invention provides a speech signal processing system of electronic communication equipment, comprising:
the voice signal acquisition unit is used for acquiring a voice signal of an information source and converting the voice signal into a setting signal;
the voice signal processing unit is used for grouping the set signals according to the channel bandwidth lower limit threshold value, processing at least two groups of voice signals with different bandwidths in the set signals, generating harmonic signals of a group of voice signals with a lower bandwidth threshold value, and overlapping the harmonic signals with a group of voice signals with a higher bandwidth threshold value to generate secondary set signals;
and the voice signal pickup unit is used for picking up the secondary setting signal and outputting the secondary setting signal through a channel.
Further, the signal processing unit includes:
the first filtering unit is used for extracting a first voice signal and a second voice signal in the setting signal according to a channel bandwidth lower limit threshold, wherein the upper limit threshold of the first voice signal bandwidth is equal to the channel lower limit threshold, and the lower limit threshold of the second voice signal bandwidth is equal to the upper limit threshold of the first voice signal bandwidth;
a harmonic generation unit for generating a harmonic signal from a fundamental frequency component of the first speech signal;
the second filtering unit is used for filtering the harmonic signals;
and the first adder is used for superposing the second voice signal and the first voice signal output by the second filtering unit and generating a secondary setting signal.
Preferably, the first filtering unit includes a band pass filter for extracting the first voice signal and a high pass filter for extracting the second voice signal.
Further, the harmonic generation unit includes:
and the frequency adjusting module is used for generating and outputting a plurality of harmonic component signals according to the fundamental frequency component of the first voice signal.
Preferably, the frequency adjustment module is a multiplier.
Further, the harmonic generation unit further includes:
a second adder for superimposing a fundamental frequency component and a plurality of harmonic components of the first voice signal and outputting a harmonic signal;
and the amplitude adjusting module is used for enabling the amplitude of the harmonic signal to be equal to or less than a set threshold value.
Further, the amplitude adjustment module causes the amplitude of the harmonic signal to be proportionally reduced when the amplitude of the harmonic signal is greater than a set threshold.
In the invention, the voice signal processing unit arranged at the sending end of the mobile communication electronic equipment divides the transmitted voice signals into two groups according to the lower limit threshold of the channel bandwidth, generates a harmonic signal taking a group of voice signals filtered by the channel as a fundamental frequency signal, and superposes the harmonic signal and a group of voice signals with bandwidth thresholds higher than the lower limit threshold of the channel bandwidth, obtains voice information with complete voice signals at the sending end and outputs the voice information through the voice signal pickup unit and the channel, and the bandwidth of the voice signals is expanded, so that the quality of the transmitted voice signals at the sending end is improved.
The invention also discloses a method for processing the voice signal of the electronic communication equipment, which comprises the following steps:
collecting a voice signal from an information source and converting the voice signal into a set signal;
grouping set signals according to a channel bandwidth lower limit threshold, processing at least two groups of voice signals with different bandwidths in the set signals to generate harmonic signals of a group of voice signals with a lower bandwidth threshold, and overlapping the harmonic signals with a group of voice signals with a higher bandwidth threshold to generate secondary set signals;
and picking up the secondary setting signal and outputting the secondary setting signal through a channel.
Further, the method also comprises the following steps:
extracting a first voice signal and a second voice signal in a set signal according to a channel bandwidth lower limit threshold, wherein the upper limit threshold of the first voice signal bandwidth is equal to the channel lower limit threshold, and the lower limit threshold of the second voice signal bandwidth is equal to the upper limit threshold of the first voice signal bandwidth;
generating a harmonic signal according to a fundamental frequency component of the first voice signal;
filtering the harmonic signals;
and superposing the second voice signal and the first voice signal output by the second filtering unit to generate a secondary setting signal.
The voice signal processing method of the electronic equipment provided by the invention compensates the signal at the signal sending end, stably realizes the improvement of the tone quality, is not influenced by the compensation algorithm of the receiving end, and has better tone quality improvement effect compared with the prior art.
The invention further discloses an electronic communication device, comprising an electronic device voice signal processing system, wherein the signal processing system comprises:
the voice signal acquisition unit is used for acquiring a voice signal of an information source and converting the voice signal into a setting signal;
the voice signal processing unit is used for grouping the set signals according to the channel bandwidth lower limit threshold value, processing at least two groups of voice signals with different bandwidths in the set signals, generating harmonic signals of a group of voice signals with a lower bandwidth threshold value, and overlapping the harmonic signals with a group of voice signals with a higher bandwidth threshold value to generate secondary set signals;
and the voice signal pickup unit is used for picking up the secondary setting signal and outputting the secondary setting signal through a channel.
The electronic equipment provided by the invention has the advantage of good communication tone quality.
Drawings
In order to more clearly illustrate the embodiments of the present invention or the technical solutions in the prior art, the drawings needed to be used in the description of the embodiments or the prior art will be briefly introduced below, and it is obvious that the drawings in the following description are some embodiments of the present invention, and for those skilled in the art, other drawings can be obtained according to these drawings without creative efforts.
FIG. 1 is a block diagram schematically illustrating a speech signal processing system of an electronic communication device according to the present invention;
FIG. 2 is a block diagram illustrating the structure of an embodiment of the speech signal processing system of the electronic communication device shown in FIG. 1;
FIG. 3 is a block diagram of a specific embodiment of the harmonic generation unit of FIG. 2;
fig. 4 is a block diagram schematically illustrating the structure of another embodiment of the harmonic generation unit in fig. 2.
Detailed Description
The invention provides a voice signal processing system of electronic communication equipment, aiming at compensating a voice signal filtered by channel bandwidth in a voice signal at a transmitting end of the electronic communication equipment. According to psychoacoustics, although fundamental frequency components in a speech signal are lost, if harmonic components of the fundamental frequency signal are contained in the speech signal, human ears can distinguish the fundamental frequency components from the harmonic components. The voice signal processing system of the electronic communication equipment provided by the invention also utilizes the principle to compensate the voice signal at the sending end of the electronic communication equipment, particularly mobile communication equipment such as a mobile phone, an intelligent watch and the like, thereby processing the voice signal below 300Hz and improving the conversation quality. Specifically, as shown in fig. 1, the voice signal processing system of the electronic communication device is composed of a voice signal collecting unit 1, a voice signal processing unit 2 and a voice signal picking unit 3. The voice signal collecting unit 1 is actually an audio signal collecting unit, and converts a voice signal into an electric signal and outputs the electric signal. The voice signal processing unit 2 is connected behind the voice signal collecting unit 1, and the electric signal converted from the voice signal can be regarded as the input setting signal of the voice signal processing unit 2. The voice signal processing unit 2 can perform at least three functions, one of which is to group the setting signals according to the channel bandwidth lower limit threshold, that is, to divide the setting signals into a group higher than the channel bandwidth lower limit threshold and a group lower than the channel bandwidth lower limit threshold. The second is that harmonic signals based on fundamental frequency components of a group of speech signals below the lower limit of the channel bandwidth are generated. And thirdly, superposing the harmonic signal and a group of voice signals which are higher than the lower limit threshold of the channel bandwidth.
In the electronic communication equipment in the prior art, a set signal can be directly transmitted through a channel by a control chip, signal compensation is carried out at a receiving end, and compensation effects are different due to different algorithms and are difficult to control. To avoid this, in the present invention, a group of signals lower than the lower threshold of the channel bandwidth is directly processed by the speech signal processing unit 2 to generate harmonic signals, and the generated harmonic signals are further superimposed with a group of signals higher than the lower threshold of the channel bandwidth to generate secondary setting signals. The voice signal pickup unit 3 in the voice signal processing system picks up the second setting signal, and the second setting signal at this time includes the harmonic component in the voice signal that has been filtered out, so according to psychoacoustics, the human ear can certainly distinguish the fundamental frequency part of the harmonic component at the receiving end, the signal quality of the source is greatly improved, the quality of the voice call is improved accordingly, and the phenomenon of sound distortion is avoided.
In order to make the objects, technical solutions and advantages of the embodiments of the present invention clearer, the technical solutions in the embodiments of the present invention will be clearly and completely described below with reference to the drawings in the embodiments of the present invention, and it is obvious that the described embodiments are some, but not all, embodiments of the present invention. All other embodiments, which can be derived by a person skilled in the art from the embodiments given herein without making any creative effort, shall fall within the protection scope of the present invention.
Fig. 2 specifically shows a functional schematic block diagram of each unit in the speech signal processing system of the electronic communication device shown in fig. 1, and particularly embodies the signal transmission and processing flow of each unit in the speech signal processing unit 2. Specifically, the setting signal transmitted to the voice signal processing unit 2 is first subjected to filtering processing. The bandwidth range of the conventional voice communication frequency is 300Hz to 3400Hz, so the first filtering unit 21 extracts the first voice signal and the second voice signal in the setting signal according to the lower limit value of 300 Hz. Wherein the upper threshold of the first voice signal bandwidth is equal to the lower threshold of the channel, that is, 300Hz, and the lower threshold of the second voice signal bandwidth is equal to the upper threshold of the first voice signal, that is, the voice signal with frequency higher than 300Hz transmitted through the channel in the conventional communication. The filtered voice signal satisfies its own waveform and frequency characteristics. The first filtering unit 21 specifically preferably includes a band-pass filter 21-1(150Hz to 300Hz) and a high-pass filter 21-2(300Hz), the band-pass filter 21-1 can obtain the fundamental frequency components of the part of the voice signal filtered by the transmitting end of the conventional communication device, and similarly, the high-pass filter 21-2 can obtain the voice signal normally transmitted by the transmitting end of the conventional communication device, i.e., the second voice signal.
The speech signal processing unit 2 further comprises a harmonic generation unit 22 to generate harmonic signals having frequencies that are integer multiples of the fundamental frequency component. Fig. 3 is a schematic structural diagram of an embodiment of the harmonic generation unit, which is also a simplest implementation form of the harmonic generation unit 22 that can be used in the present invention. In this embodiment, the harmonic generation unit 22 at least includes a frequency adjustment module 22-1, which is implemented by a multiplier and has at least two signal input channels 22-2, and each signal input channel 22-2 inputs a fundamental frequency component of the first speech signal generated by filtering with the band-pass filter 21-1, which is defined as X ═ Ae-j2πftThe fundamental frequency is f, so that the output signal after the multiplier is Xout＝A2e-j2π2ftThen, a 2 nd harmonic component of the fundamental frequency component of the first speech signal with the frequency 2f is obtained. In the same way, by adjusting the number of signal input channels of the harmonic generation unit, 3 rd harmonic component, 4 th harmonic component, and the like of the fundamental frequency component of the first speech signal with frequencies 3f, 4f can be obtained. The ideal harmonic components are added by an adder, so that a harmonic signal generated according to the fundamental frequency component of the first speech signal can be obtained.
As shown in fig. 2, a second filtering unit 23 is further disposed downstream of the harmonic generation unit 22-1 of the speech signal processing unit 2, the second filtering unit 23 is preferably a high-pass filter, the setting value is preferably a lower threshold of a channel bandwidth, in this embodiment, 300Hz, so as to further filter the harmonic signal by the second filtering unit 23 to meet the frequency characteristics of channel transmission.
The harmonic signal input from the second filtering unit 23 and the second voice signal are superimposed by the first adder 24, and the second-order setting signal received by the voice signal pickup unit 3 is generated.
The voice signal pickup unit 3 receives the secondary setting signal and then transmits the secondary setting signal through a channel, thereby realizing spread spectrum voice communication. The voice signal pick-up unit 3 may be part of a control chip or may be a separate chip or integrated circuit in the electronic communication device that may implement this function.
In this embodiment, a voice signal processing unit disposed at a sending end of a mobile communication electronic device divides transmitted voice signals into two groups according to a channel bandwidth lower limit threshold, generates a harmonic signal using a group of voice signals filtered by a channel as a fundamental frequency signal, and superimposes the harmonic signal and a group of voice signals in which the bandwidth threshold is higher than the channel bandwidth lower limit threshold, so that voice information with complete voice signals is obtained at the sending end and is output through a voice signal pickup unit and a channel.
Fig. 4 shows a second specific embodiment of the harmonic generation unit 22 used in the speech signal processing unit according to the present invention. Wherein, the harmonic generation unit 22 includes a frequency adjustment module 22-1, and the frequency adjustment module 22-1 is also implemented by a multiplier. The frequency adjustment module 22-1 has at least one signal input channel 22-2. Wherein the fundamental frequency component of the input first speech signal is denoted as X ═ Ae-j2πft. The harmonic generation unit 22 further includes a second adder 22-4, the second adder 22-4 has at least two signal input channels, wherein one input signal is a voice signal output by the frequency adjustment module 22-1, the other input signal is a fundamental frequency component in the first voice signal, and the signal superimposed by the second adder 22-4 is returned to one signal input channel 22-5 of the frequency adjustment module 22-1 as a feedback signal, and is combined with the frequencyThe fundamental frequency component in the signal input channel 22-2 of the rate adjustment module 22-1 is multiplied in the multiplier and then output to form the 2 nd harmonic component with the frequency 2f, and the 2 nd harmonic component is again superimposed with the fundamental frequency component in the first voice signal in the second adder 22-4 and is returned to the frequency adjustment module 22-1 as a feedback signal. In this way, harmonic components with frequencies that are ideal multiples are obtained. In principle, the richer the harmonic component is superimposed into the second speech signal, the higher the quality of the speech signal, and the better the restoration degree of the speech signal tone.
In the harmonic generation unit disclosed in fig. 4, the amplitude of the signal multiplied by the accumulation is also increased, which gradually increases the intensity of the part of the speech signal that is originally filtered out. However, the amplitude, and more precisely, the absolute value of the amplitude, increases all the time, and distortion of the speech signal also occurs. Therefore, the amplitude adjusting module 22-6 is arranged on the output end of the second adder 22-4 and the feedback loop of the frequency adjusting module 22-1, and the amplitude adjusting module 22-6 enables the signal output by the second adder 22-4 and fed back to the frequency adjusting module to be multiplied by a coefficient which is larger than 0 and smaller than 1, and then fed back to the frequency adjusting module 22-1, so that the amplitude of the harmonic signal does not exceed the set threshold value, and the phenomenon of voice signal distortion is avoided. The reduced amplitude portion is compensated for when the second adder 22-4 is added, effectively avoiding unwanted attenuation.
The invention also discloses a speech signal processing method of the electronic communication equipment, which comprises the following steps:
firstly, a voice signal of an information source is collected and converted into a setting signal.
Secondly, grouping the set signals according to the lower limit threshold of the channel bandwidth, processing at least two groups of voice signals with different bandwidths in the set signals to generate harmonic signals of a group of voice signals with a lower bandwidth threshold, and overlapping the harmonic signals with a group of voice signals with a higher bandwidth threshold to generate secondary set signals.
Thirdly, the secondary setting signal is picked up and output through the channel.
The specific steps for converting the setting signal into a second setting signal having a harmonic component are as follows:
first, the first filtering unit extracts a first voice signal and a second voice signal in the setting signal according to a lower threshold of a channel bandwidth, wherein an upper threshold of the first voice signal bandwidth is equal to a lower threshold of the channel bandwidth, and a lower threshold of the second voice signal bandwidth is equal to an upper threshold of the first voice signal bandwidth, preferably, the first voice signal bandwidth is (150Hz to 300HHz), and the second voice signal bandwidth is (greater than 300 Hz).
Second, a harmonic signal is generated from the fundamental frequency component of the first speech signal. For the method for generating the harmonic signal from the first speech signal, reference may be made to the detailed description of the harmonic generating unit in the first embodiment and the second embodiment and fig. 3 and 4, which are not repeated herein.
Thirdly, the generated harmonic signal is subjected to filtering processing.
And fourthly, superposing the second voice signal and the first voice signal output by the second filtering unit and generating a secondary setting signal.
By adopting the speech signal processing method of the electronic communication equipment disclosed by the invention, the speech information with complete speech signals is obtained at the sending end and is output through the speech signal pick-up unit and the channel, and the bandwidth of the speech signals is expanded, so that the quality of the transmitted speech signals is improved at the sending end,
meanwhile, the invention also discloses an electronic communication device which can be a mobile or fixed device with a communication function, such as a mobile phone, a mobile terminal, an intelligent watch and the like. The structure, material, and shape of the electronic communication device are not limited herein. The electronic communication device has therein the electronic communication device voice signal processing system as disclosed in the first and second embodiments. For specific technical contents of the signal processing system, please refer to detailed descriptions of the first embodiment and the second embodiment and detailed descriptions of fig. 1 to 4 in the specification, which are not repeated again. An electronic device with a signal processing system may achieve the same technical effect.
Finally, it should be noted that: the above examples are only intended to illustrate the technical solution of the present invention, but not to limit it; although the present invention has been described in detail with reference to the foregoing embodiments, it will be understood by those of ordinary skill in the art that: the technical solutions described in the foregoing embodiments may still be modified, or some technical features may be equivalently replaced; and such modifications or substitutions do not depart from the spirit and scope of the corresponding technical solutions of the embodiments of the present invention.
Claims (10)
1. A speech signal processing system for an electronic communication device, comprising
The voice signal acquisition unit is used for acquiring a voice signal of an information source and converting the voice signal into a setting signal;
the voice signal processing unit is used for grouping the set signals according to the channel bandwidth lower limit threshold to obtain a first voice signal and a second voice signal, processing the first voice signal to generate a harmonic signal, and overlapping the harmonic signal and the second voice signal to generate a secondary set signal; wherein, the upper threshold of the first voice signal bandwidth is equal to the lower threshold of the channel, and the lower threshold of the second voice signal bandwidth is equal to the upper threshold of the first voice signal bandwidth;
and the voice signal pickup unit is used for picking up the secondary setting signal and outputting the secondary setting signal through a channel.
2. The electronic communication device voice signal processing system of claim 1, wherein the signal processing unit comprises:
the first filtering unit is used for extracting a first voice signal and a second voice signal in the set signal according to the lower limit threshold of the channel bandwidth;
a harmonic generation unit for generating a harmonic signal from a fundamental frequency component of the first speech signal;
the second filtering unit is used for filtering the harmonic signals;
and the first adder is used for superposing the second voice signal and the first voice signal output by the second filtering unit and generating a secondary setting signal.
3. The electronic communication device voice signal processing system of claim 2, wherein the first filtering unit comprises a band-pass filter for extracting the first voice signal and a high-pass filter for extracting the second voice signal.
4. The electronic communication device voice signal processing system of claim 3, wherein the harmonic generation unit comprises:
and the frequency adjusting module is used for generating and outputting a plurality of harmonic component signals according to the fundamental frequency component of the first voice signal.
5. The electronic communication device voice signal processing system of claim 4, wherein the frequency adjustment module is a multiplier.
6. The electronic communication device voice signal processing system of claim 5, wherein the harmonic generation unit further comprises:
a second adder for superimposing a fundamental frequency component and a plurality of harmonic components of the first voice signal and outputting a harmonic signal;
and the amplitude adjusting module is used for enabling the amplitude of the harmonic signal to be equal to or less than a set threshold value.
7. The electronic communication device voice signal processing system of claim 6, wherein the amplitude adjustment module causes the amplitude of the harmonic signal to be proportionally reduced when the amplitude of the harmonic signal is greater than a set threshold.
8. A speech signal processing method of an electronic communication device, characterized by:
the method comprises the following steps:
collecting a voice signal from an information source and converting the voice signal into a set signal;
grouping the set signals according to a channel bandwidth lower limit threshold to obtain a first voice signal and a second voice signal, processing the first voice signal to generate a harmonic signal, and overlapping the harmonic signal and the second voice signal to generate a secondary set signal; wherein the upper bandwidth threshold of the first voice signal is equal to the lower channel threshold, and the lower bandwidth threshold of the second voice signal is equal to the upper bandwidth threshold of the first voice signal;
and picking up the secondary setting signal and outputting the secondary setting signal through a channel.
9. The electronic communication device voice signal processing method of claim 8, further comprising the steps of:
extracting a first voice signal and a second voice signal in a set signal according to a channel bandwidth lower limit threshold;
generating a harmonic signal according to a fundamental frequency component of the first voice signal;
filtering the harmonic signals;
and superposing the second voice signal and the harmonic signal after filtering to generate a secondary setting signal.
10. An electronic communication device comprising an electronic communication device speech signal processing system according to any of claims 1 to 7.
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| CN1630427A (en) * | 2004-06-30 | 2005-06-22 | 深圳兰光电子集团有限公司 | A method of bass boosting processing |
| CN101512639A (en) * | 2006-09-13 | 2009-08-19 | 艾利森电话股份有限公司 | Method and equipment for voice/audio transmitter and receiver |
| JP5291577B2 (en) * | 2009-08-31 | 2013-09-18 | アイホン株式会社 | Intercom system |
| CN102752068A (en) * | 2011-04-22 | 2012-10-24 | 松下电器产业株式会社 | Communication system, transmitting apparatus, receiving apparatus and communication method |
| CN102932710A (en) * | 2012-10-10 | 2013-02-13 | 广州创维平面显示科技有限公司 | Method and device for bass compensation |
| CN103928029A (en) * | 2013-01-11 | 2014-07-16 | 华为技术有限公司 | Audio signal coding method, audio signal decoding method, audio signal coding apparatus, and audio signal decoding apparatus |
| CN105830153A (en) * | 2013-12-16 | 2016-08-03 | 高通股份有限公司 | High-band signal modeling |
| CN105632509A (en) * | 2014-11-07 | 2016-06-01 | Tcl集团股份有限公司 | Audio processing method and audio processing device |
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