CN104158942A - Voice processing device - Google Patents

Abstract

The invention provides a voice processing device. The voice processing device comprises an audio input device and an audio output device, wherein the audio input device is connected to a loudspeaker through an audio amplifying circuit; the loudspeaker is used for outputting a voice signal received from another terminal device to a user; and the audio output device is used for receiving the voice signal of the user, and transmitting the voice signal to another terminal device through a mobile communication network. According to the voice processing device provided by the invention, power compensation is performed on an audio signal input into the audio input device through a compensating circuit, and the volume of the loudspeaker is adjusted automatically according to a volume differential value between received and output audio signals, so that comfortable conversation experience is brought to the user, setting of volume keys is not needed, and the compactness and attractiveness of a mobile phone are further improved; and audio signals of a master MIC (Microphone) and a slave MIC are amplified and attenuated through a filtering and optimizing module, so that the transmission rate of environmental noise is reduced greatly, and the listening experience of the user is improved.

Description

A kind of voice processing apparatus

Technical field

The present invention relates to electronic applications, refer to especially a kind of voice processing apparatus.

Background technology

Existing mobile phone is divided into smart mobile phone and non intelligent mobile phone, and the non intelligent mobile phone of Performance Ratio of general intelligence mobile phone is good, but non intelligent mobile phone is more stable than smart mobile phone, and most of non intelligent mobile phones and smart mobile phone use the CPU of Britain ARM corporate structure.The dominant frequency of smart mobile phone is higher, and the speed of service is fast, and smart mobile phone handling procedure task is quicker, daily more convenient, but not the dominant frequency of smart mobile phone is lower, and the speed of service is also slow.

No matter be smart mobile phone or non intelligent mobile phone, the control of its volume is mainly the adjusting by manually realizing volume, and smart mobile phone is realized volume adjusting by real key or touch-screen virtual key, and non intelligent mobile phone regulates volume by real key, at present, also do not have the mobile phone of automatic regulating volume to occur.And its tonequality of existing mobile phone is all accomplished the effect of high-fidelity as far as possible, i.e. former sound transmission, but for the noisy occasion such as public transport, market, the user of the other end often audibility is very poor, and user experiences greatly and reduces.

Summary of the invention

The present invention proposes a kind of voice processing apparatus, has solved in prior art by button and has realized volume adjusting and noise by the problem of synchronous transmission.

Technical scheme of the present invention is achieved in that

A kind of voice processing apparatus, comprise: voice input device and audio output device, described voice input device is connected to loud speaker by audio amplifier circuit, loud speaker is exported the voice signal of another terminal equipment receiving to user, audio output device receives user's voice signal and it is transferred to another terminal equipment by mobile communications network

Described voice input device is also connected to the first sample circuit, and the first sample circuit gathers the volume value of voice input device and is converted into corresponding the first magnitude of voltage;

Described audio output device is connected to the second sample circuit, and the second sample circuit gathers the volume value of audio output device and is converted into corresponding second voltage value;

The first comparator is hysteresis comparator, and the first magnitude of voltage and second voltage value are compared, and in the time that the difference of the first magnitude of voltage and second voltage value exceeds preset range, the first comparator output difference threshold voltage is to the first compensating circuit;

The first compensating circuit is on-off controller, according to the adjustable pulse signal of described difference voltage output duty cycle, controls turning on and off of switching tube in audio amplifier circuit, and regulation output is to the power of loud speaker;

Described audio output device comprises:.

Be arranged on the main MIC of mobile phone front below and be arranged on top, the mobile phone back side from MIC;

Filtering and optimization module comprise the first subfilter, the second subfilter and senior filter; The first subfilter is processed main MIC voice data information, and output state is estimated x1 and estimation error covariance matrix p1; The second subfilter is processed from MIC voice data information, and output state is estimated x2 and estimation error covariance matrix p2; Described state estimation x1, x2 and estimation error covariance matrix p1, p2 delivers to senior filter, and merges by formula (1) and formula (2) together with the state estimation of senior filter, obtains global optimum and estimates and covariance matrix:

${\hat{x}}_g=p_g\left(\underset{i=1}{\overset{2}{\mathrm{\Σ}}}{p}_i^{-1}{\hat{x}}_i\right)---\left(1\right)$

$p_g=\underset{i=1}{\overset{2}{\mathrm{\Σ}}}{p}_i^{-1}---\left(2\right)$

In formula, i=1,2;

The optimal estimation value of senior filter is reset to the state estimation of 2 subfilters,

${\hat{x}}_i={\hat{x}}_g---\left(3\right)$

$p_i={\mathrm{\β}}_i^{-1}{p}_g---\left(4\right)$

Using main MIC voice data information as analyzing datum quantity, automatically adjust the size of p value, concrete adaptive algorithm is:

β ₁, β ₂meet β ₁+ β ₂=1 (5)

${\mathrm{\β}}_1=\left\{\begin{array}{c}0.99,p\≤2\\ 2/p,2\≤p\≤5\\ 1/p,5\≤p\≤10\\ 2/p,p\≥10\end{array}\right.---\left(6\right)$

Mobile communications network receives described filtering and optimizes the output signal of module, and the audio signal after optimizing is sent to another mobile reception end;

Described the first sample circuit, the second sample circuit, the first comparator, the first compensating circuit and filtering and optimization module realize by DSP circuit, described DSP circuit is TMS320DM642 chip, builds described the first sample circuit, the second sample circuit, the first comparator, the first compensating circuit and filtering and optimize module by programming in dsp chip inside;

Described audio amplifier circuit comprises:

Transformer, comprise former limit winding and secondary winding, former limit winding comprises first end and the second end, secondary winding comprises first end and the second end, former limit winding first end is connected to battery, secondary winding the second end is connected to secondary earth potential, and former limit winding first end and secondary winding the second end are Same Name of Ends;

The first switching tube, its source electrode is connected to transformer primary side winding the second end, and its drain electrode is connected to former limit earth potential, and its grid is connected to the first switch driver;

The first switch driver, comprise the first transistor and transistor seconds, the first transistor is NPN transistor, transistor seconds is PNP transistor, the base stage of the first transistor and transistor seconds links together as control end, and control end receives the pulse signal of described the first compensating circuit output, and the emitter of the first transistor and transistor seconds links together as output, the collector electrode of the first transistor is connected to described battery, and the collector electrode of transistor seconds is connected to former limit earth potential;

The first diode, its anodic bonding is to transformer secondary winding first end, and its negative electrode is connected to the first inductor;

The common port of the first inductor and the first capacitor is as output, and the other end of the first capacitor is connected to secondary earth potential.

The invention has the beneficial effects as follows:

By compensating circuit, the audio signal that is input to voice input device is carried out to power back-off, automatically regulate the volume of loud speaker according to the volume difference of the audio signal receiving and export, make user's call experience comfortable, and the setting that can omit volume key, further improve compactedness and the aesthetic property of mobile phone; Amplify and decay to main MIC with from the audio signal of MIC by filtering and optimization module, greatly reduce the transfer rate of environmental noise, improve user's the experience of listening to.

Brief description of the drawings

In order to be illustrated more clearly in the embodiment of the present invention or technical scheme of the prior art, to the accompanying drawing of required use in embodiment or description of the Prior Art be briefly described below, apparently, accompanying drawing in the following describes is only some embodiments of the present invention, for those of ordinary skill in the art, do not paying under the prerequisite of creative work, can also obtain according to these accompanying drawings other accompanying drawing.

Fig. 1 is the control block diagram of voice processing apparatus of the present invention;

Fig. 2 is filtering of the present invention and the control block diagram of optimizing module;

Fig. 3 is the circuit diagram of audio amplifier circuit of the present invention.

Embodiment

Below in conjunction with the accompanying drawing in the embodiment of the present invention, the technical scheme in the embodiment of the present invention is clearly and completely described, obviously, described embodiment is only the present invention's part embodiment, instead of whole embodiment.Based on the embodiment in the present invention, those of ordinary skill in the art, not making the every other embodiment obtaining under creative work prerequisite, belong to the scope of protection of the invention.

As shown in Figure 1, voice processing apparatus of the present invention, comprise: voice input device 100 and audio output device 400, described voice input device 100 is connected to loud speaker 800 by audio amplifier circuit 700, loud speaker 800 is exported the voice signal of another terminal equipment receiving to user, audio output device 400 receives user's voice signal and it is transferred to another terminal equipment by mobile communications network, described voice input device 100 is also connected to the first sample circuit 200, the first sample circuit 200 gathers the volume value of voice input device 100 and is converted into corresponding the first magnitude of voltage, described audio output device 400 is connected to the second sample circuit 300, the second sample circuits 300 and gathers the volume value of audio output device and be converted into corresponding second voltage value, the first comparator 500 is hysteresis comparator, and the first magnitude of voltage and second voltage value are compared, and in the time that the difference of the first magnitude of voltage and second voltage value exceeds preset range, the first comparator output difference threshold voltage is to the first compensating circuit 600, the first compensating circuit 600 is on-off controller, according to the adjustable pulse signal of described difference voltage output duty cycle, controls turning on and off of switching tube in audio amplifier circuit 700, and regulation output is to the power of loud speaker.

As shown in Figure 2, audio output device 400 of the present invention, comprising: be arranged on the main MIC110 of mobile phone front below and be arranged on top, the mobile phone back side from MIC120.

Filtering and optimization module 130 comprise the first subfilter 1311, the second subfilter 1312 and senior filter 1314; The first subfilter is processed main MIC voice data information, and output state is estimated x1 and estimation error covariance matrix p1; The second subfilter is processed from MIC voice data information, and output state is estimated x2 and estimation error covariance matrix p2; Described state estimation x1, x2 and estimation error covariance matrix p1, p2 delivers to senior filter, and merges by formula (1) and formula (2) together with the state estimation of senior filter, obtains global optimum and estimates and covariance matrix:

${\hat{x}}_g=p_g\left(\underset{i=1}{\overset{2}{\mathrm{\Σ}}}{p}_i^{-1}{\hat{x}}_i\right)---\left(1\right)$

$p_g=\underset{i=1}{\overset{2}{\mathrm{\Σ}}}{p}_i^{-1}---\left(2\right)$

In formula, i=1,2;

The optimal estimation value of senior filter is reset to the state estimation of 2 subfilters,

${\hat{x}}_i={\hat{x}}_g---\left(3\right)$

$p_i={\mathrm{\β}}_i^{-1}{p}_g---\left(4\right)$

Using main MIC voice data information as analyzing datum quantity, automatically adjust the size of p value, concrete adaptive algorithm is:

β ₁, β ₂meet β ₁+ β ₂=1 (5)

${\mathrm{\β}}_1=\left\{\begin{array}{c}0.99,p\≤2\\ 2/p,2\≤p\≤5\\ 1/p,5\≤p\≤10\\ 2/p,p\≥10\end{array}\right.---\left(6\right)$

Mobile communications network 810 receives described filtering and optimizes the output signal of module 130, and the audio signal after optimizing is sent to another mobile reception end.

Described the first sample circuit, the second sample circuit, the first comparator, the first compensating circuit and filtering and optimization module can realize by discrete form, also can realize by integrated form, discrete form i.e. the first sample circuit, the second sample circuit, the first comparator, the first compensating circuit and filtering and optimization module realize by multiple discrete chips and peripheral component thereof, integrated form sets up the first sample circuit by the programming of DSP circuit in dsp chip inside, the second sample circuit, the first comparator, the first compensating circuit and filtering and optimization module, described DSP circuit is TMS320DM642 chip.

TMS320DM642 is up-to-date fixed DSP in the C6000 of TI company series DSP, its core is C6416 type high-performance digital signal processor, there is extremely strong handling property, flexibility and the programmability of height, equipment and the interfaces such as simultaneously peripheral integrated very complete audio frequency, video and network service.

TMS320DM642 chip comprises: 3 configurable video ports (VPORT0-2) can be realized seamless link with general video coding and decoding device, support various video resolution and video standard, support RAW video input/output, transport stream mode; 1 10/100Mb/s Ethernet interface (EMAC), meets IEEE802.3 standard; 1 multichannel band buffered audio serial port (McASP), supports I2S, DIT, S/PDIF, IEC60958-1, the audio formats such as AES-3, CP-430; 2 multichannel band buffering serial ports (McBSP), adopt RS232 level to drive; 1 VCXO interpolation control unit (VIC), supports that audio/video is synchronous; 1 32,66M hertz, the master/slave pci interface of 3.3V, follow PCI2.2 specification; 1 configurable 16/32 host interface of user (HPI); 1 16 general input/output port (GPIO); 1 64 external memory interfaces (EMIF), can with most of asynchronous memories (SRAM, EPROM) and synchronous memories (SDRAM, SBSRAM, ZBTSRAM, FIFO) seamless link, maximum addressable external memory storage space is 1024MB; 1 has the enhancement mode direct memory access controller (EDMA) of 64 autonomous channels, tunnel; 1 data management input/output module (MDIO); 1 I2C bus module; 3 32 general purpose timers; 1 meets jtag interface and the daughter board interface etc. of IEEE1149.1 standard.

As shown in Figure 3, audio amplifier circuit of the present invention comprises: transformer 51, comprise former limit winding and secondary winding, former limit winding comprises first end and the second end, secondary winding comprises first end and the second end, former limit winding first end is connected to battery as input, and secondary winding the second end is connected to secondary earth potential, and former limit winding first end and secondary winding the second end are Same Name of Ends; The first switching tube 52, its source electrode is connected to transformer primary side winding the second end, and its drain electrode is connected to former limit earth potential, and its grid is connected to the first switch driver 53; The first switch driver 53, comprise the first transistor and transistor seconds, the first transistor is NPN transistor, transistor seconds is PNP transistor, the base stage of the first transistor and transistor seconds links together as control end, the emitter of the first transistor and transistor seconds links together as output, and the collector electrode of the first transistor is connected to battery, and the collector electrode of transistor seconds is connected to former limit earth potential; The first diode 54, its anodic bonding is to transformer secondary winding first end, and its negative electrode is connected to the first inductor 55; The common port of the first inductor 55 and the first capacitor 56 is as output, and the other end of the first capacitor 56 is connected to secondary earth potential.

The impulse circuit of described the first compensating circuit output is couple to the control end of the first switch driver 53, the turn-on and turn-off of control switch pipe.

Voice processing apparatus of the present invention, by compensating circuit, the audio signal that is input to voice input device is carried out to power back-off, automatically regulate the volume of loud speaker according to the volume difference of the audio signal receiving and export, make user's call experience comfortable, and the setting that can omit volume key, further improve compactedness and the aesthetic property of mobile phone; Amplify and decay to main MIC with from the audio signal of MIC by filtering and optimization module, greatly reduce the transfer rate of environmental noise, improve user's the experience of listening to.

The foregoing is only preferred embodiment of the present invention, in order to limit the present invention, within the spirit and principles in the present invention not all, any amendment of doing, be equal to replacement, improvement etc., within all should being included in protection scope of the present invention.

Claims (1)

1. a voice processing apparatus, comprise: voice input device and audio output device, described voice input device is connected to loud speaker by audio amplifier circuit, loud speaker is exported the voice signal of another terminal equipment receiving to user, audio output device receives user's voice signal and it is transferred to another terminal equipment by mobile communications network, it is characterized in that

Described voice input device is also connected to the first sample circuit, and the first sample circuit gathers the volume value of voice input device and is converted into corresponding the first magnitude of voltage;

Described audio output device is connected to the second sample circuit, and the second sample circuit gathers the volume value of audio output device and is converted into corresponding second voltage value;

The first comparator is hysteresis comparator, and the first magnitude of voltage and second voltage value are compared, and in the time that the difference of the first magnitude of voltage and second voltage value exceeds preset range, the first comparator output difference threshold voltage is to the first compensating circuit;

The first compensating circuit is on-off controller, according to the adjustable pulse signal of described difference voltage output duty cycle, controls turning on and off of switching tube in audio amplifier circuit, and regulation output is to the power of loud speaker;

Described audio output device comprises:

Be arranged on the main MIC of mobile phone front below and be arranged on top, the mobile phone back side from MIC;

Filtering and optimization module comprise the first subfilter, the second subfilter and senior filter; The first subfilter is processed main MIC voice data information, and output state is estimated x1 and estimation error covariance matrix p1; The second subfilter is processed from MIC voice data information, and output state is estimated x2 and estimation error covariance matrix p2; Described state estimation x1, x2 and estimation error covariance matrix p1, p2 delivers to senior filter, and merges by formula (1) and formula (2) together with the state estimation of senior filter, obtains global optimum and estimates and covariance matrix:

${\hat{x}}_g=p_g\left(\underset{i=1}{\overset{2}{\mathrm{\Σ}}}{p}_i^{-1}{\hat{x}}_i\right)---\left(1\right)$

$p_g=\underset{i=1}{\overset{2}{\mathrm{\Σ}}}{p}_i^{-1}---\left(2\right)$

In formula, i=1,2;

The optimal estimation value of senior filter is reset to the state estimation of 2 subfilters,

${\hat{x}}_i={\hat{x}}_g---\left(3\right)$

$p_i={\mathrm{\β}}_i^{-1}{p}_g---\left(4\right)$

Using main MIC voice data information as analyzing datum quantity, automatically adjust the size of p value, concrete adaptive algorithm is:

β ₁, β ₂meet β ₁+ β ₂=1 (5)

${\mathrm{\β}}_1=\left\{\begin{array}{c}0.99,p\≤2\\ 2/p,2\≤p\≤5\\ 1/p,5\≤p\≤10\\ 2/p,p\≥10\end{array}\right.---\left(6\right)$

Mobile communications network receives described filtering and optimizes the output signal of module, and the audio signal after optimizing is sent to another mobile reception end;

Described the first sample circuit, the second sample circuit, the first comparator, the first compensating circuit and filtering and optimization module realize by DSP circuit, described DSP circuit is TMS320DM642 chip, builds described the first sample circuit, the second sample circuit, the first comparator, the first compensating circuit and filtering and optimize module by programming in dsp chip inside;

Described audio amplifier circuit comprises:

Transformer, comprise former limit winding and secondary winding, former limit winding comprises first end and the second end, secondary winding comprises first end and the second end, former limit winding first end is connected to battery, secondary winding the second end is connected to secondary earth potential, and former limit winding first end and secondary winding the second end are Same Name of Ends;

The first switching tube, its source electrode is connected to transformer primary side winding the second end, and its drain electrode is connected to former limit earth potential, and its grid is connected to the first switch driver;

The first switch driver, comprise the first transistor and transistor seconds, the first transistor is NPN transistor, transistor seconds is PNP transistor, the base stage of the first transistor and transistor seconds links together as control end, and control end receives the pulse signal of described the first compensating circuit output, and the emitter of the first transistor and transistor seconds links together as output, the collector electrode of the first transistor is connected to described battery, and the collector electrode of transistor seconds is connected to former limit earth potential;

The first diode, its anodic bonding is to transformer secondary winding first end, and its negative electrode is connected to the first inductor;

The common port of the first inductor and the first capacitor is as output, and the other end of the first capacitor is connected to secondary earth potential.