CN101211563B - sound signal processing method - Google Patents
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- CN101211563B CN101211563B CN2006101722547A CN200610172254A CN101211563B CN 101211563 B CN101211563 B CN 101211563B CN 2006101722547 A CN2006101722547 A CN 2006101722547A CN 200610172254 A CN200610172254 A CN 200610172254A CN 101211563 B CN101211563 B CN 101211563B
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- 230000005236 sound signal Effects 0.000 title claims abstract description 17
- 238000003672 processing method Methods 0.000 title claims abstract description 13
- 238000005070 sampling Methods 0.000 claims abstract description 70
- 238000000034 method Methods 0.000 claims description 18
- 238000001914 filtration Methods 0.000 claims description 9
- 238000010586 diagram Methods 0.000 description 12
- 238000000926 separation method Methods 0.000 description 11
- 238000009434 installation Methods 0.000 description 7
- 230000008569 process Effects 0.000 description 4
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Abstract
一种声音信号处理方法,包括下列步骤:首先,根据预设声音的音量,得到电压信号;接着,根据电压信号的最小电压峰值及最大电压峰值定义音量取样区间;然后,根据音量取样区间,对待取样声音进行取样。
A sound signal processing method includes the following steps: first, obtaining a voltage signal according to a preset sound volume; then, defining a volume sampling interval according to a minimum voltage peak value and a maximum voltage peak value of the voltage signal; and then, sampling the sound to be sampled according to the volume sampling interval.
Description
Technical field
The present invention relates to a kind of sound signal processing method, and be particularly related to a kind of according to the sound signal processing method of different source of sound features with the adjustment sampling mode.
Background technology
The present mobile phone of selling on the market, the design that part has had the noise resistance technology, volume in the environment is excessive or frequency is too high and low excessively sound filtering, user's sound can clearly be sent out.But present existing technology is audible frequency and the volume at general masses to be designed, for the special masses of part sounding, the loud and sonorous women or the deep-voiced male sex of sound for example, suffer filtering because their audible frequency is higher or lower than the sampling scope, thereby can't use this kind noise resistance function.
Summary of the invention
In view of this, purpose of the present invention just provides a kind of sound signal processing method, can keep user's sound and filtering environmental noise at the sound mark adjustment sampling scope of different people.
According to purpose of the present invention, first kind of sound signal processing method proposed, comprise the following steps: that at first the volume according to default sound obtains voltage signal; Then, minimum voltage peak value and the maximum voltage peak value according to voltage signal defines the volume sampling range; Then, according to the volume sampling range, treat sampling sound and take a sample.
According to purpose of the present invention, second kind of sound signal processing method proposed, comprise the following steps.At first, the frequency according to default sound obtains frequency signal; Then, minimum frequency peak value and the maximum frequency peak value according to frequency signal defines the frequency sample interval; Then, according to the frequency sample interval, treat sampling sound and take a sample.
According to purpose of the present invention, the third sound signal processing method is proposed, comprise the following steps.At first, the volume according to default sound obtains voltage signal; Then, minimum voltage peak value and the maximum voltage peak value according to voltage signal defines n voltage range; Then, s1 to the t1 voltage range of n voltage range is set at the volume sampling range, wherein 1<s1<t1<n; Then, according to the volume sampling range, treat sampling sound and take a sample.
According to purpose of the present invention, the 4th kind of sound signal processing method proposed, comprise the following steps.At first, the frequency according to default sound obtains frequency signal; Then, minimum frequency peak value and the maximum frequency peak value according to frequency signal defines m frequency separation; Then, s2 to the t2 frequency separation of m frequency separation is set at the frequency sample interval, wherein 1<s2<t2<m; Then, according to the frequency sample interval, treat sampling sound and take a sample.
For above-mentioned purpose of the present invention, feature and advantage can be become apparent, preferred embodiment cited below particularly, and conjunction with figs. are described in detail below:
Description of drawings
Fig. 1 represents to use the functional schematic of electronic installation of the present invention;
Fig. 2 represents the sound signal processing process flow diagram of the embodiment of the invention one;
Fig. 3 represents the voltage signal synoptic diagram of default wave volume;
Fig. 4 represent to wait to take a sample voltage signal synoptic diagram of sound;
Fig. 5 represents the frequency signal synoptic diagram of default sound;
Fig. 6 represent to wait to take a sample frequency signal synoptic diagram of sound; And
Fig. 7 represents the sound signal processing process flow diagram of the embodiment of the invention two.
The main element description of symbols
100: electronic installation
110: body
120: microphone
130: processing unit
140: mnemon
150: set button
Embodiment
Embodiment one
Please refer to Fig. 1, the functional schematic of electronic installation of the present invention is used in its expression.Electronic installation 100 comprises body 110, microphone 120, processing unit 130, mnemon 140 and sets button 150.Include user's sound via microphone 120 after, treated unit 130 is stored in the mnemon 140 after being converted into the voltage signal of representing volume and representing the frequency signal of frequency.After processing unit 130 obtains the sampling scope according to the voltage signal of representing user's sound characteristic and frequency signal, be stored in the mnemon 140, promptly user's sound is carried out filtering backward, user's sound can clearly be passed on away according to this sampling scope.And in the time will changing the user, only need press the setting button 150 that is connected with processing unit 130, and original sampling scope can be wiped, obtain new sampling scope with the sound of recording new user.Electronic installation 100 can comprise mobile phone, recording pen, language learner, MP3 player or any electronic installation with function of radio receiver.
Please refer to Fig. 2, it represents the sound signal processing process flow diagram of the embodiment of the invention one, and please also refer to the assembly label of Fig. 1.At first, shown in step 201, and please refer to Fig. 3, the voltage signal synoptic diagram of the default wave volume of its expression.Electronic installation 100 is with microphone 120 volume according to default sound, and for example user's volume of speaking under quiet environment obtains the voltage signal Vs as Fig. 3.
Then, shown in step 202, processing unit 130 is according to minimum voltage peak L V and the maximum voltage peak value HV definition volume sampling range of voltage signal Vs, just between the 0V-5V.
Then, shown in step 203, the volume sampling range is divided into n voltage range.With voltage signal Vs is example because minimum voltage peak L V and maximum voltage peak value HV between 0V and 5V, make n=8, with 0V to dividing into 8 intervals between 5V.Just 0V-0.625V is set at first interval, 0.625V-1.25V and is set at second interval, 1.25V-1.875V and is set at the 3rd interval, 1.875V-2.5V and is set at the 4th interval, 2.5V-3.125V and is set at the 5th interval, 3.125V-3.75V and is set at that the 6th interval, 3.75V-4.375V are set between SECTOR-SEVEN, 4.375V-5V is set between Section Eight, voltage signal Vs is carried out 8 coding.The scope of each voltage range is identical in the present embodiment, but also can set different scopes separately, and the present invention is not as limit.In addition, though present embodiment is done 8 coding with volume, also can higher resolution divide more sampling range, for example 24,32 or 128 or the like are to obtain finer and smoother volume change.
Then, shown in step 204, and please refer to Fig. 4, its expression wait to take a sample voltage signal synoptic diagram of sound.Processing unit 130 is according to the volume sampling range of step 203 definition, and the sampling voltage signal SV that treats that treats sampling sound takes a sample, and for example the user is in the noisy environments one's voice in speech.That is to say that processing unit 130 is treated the pairing part for the treatment of that sampling voltage signal SV is positioned at the volume sampling range of volume of sampling sound and taken a sample.Since the volume sampling range be positioned at 0V-5V between, therefore can not take a sample greater than the voltage signal of 5V, and obtain the sound that volume is moderate.
Then, shown in step 205, and please refer to Fig. 5, the frequency signal synoptic diagram of the default sound of its expression.Frequency according to default sound obtains frequency signal Fs.
Then, shown in step 206, according to minimum frequency peak L F and the maximum frequency peak value HF of frequency signal Fs, definition frequency sample interval is just between the 500Hz-2000Hz.
Then, shown in step 207, m frequency separation divided in the frequency sample interval.In the present embodiment, frequency signal Fs between 500Hz-2000Hz, makes m=8 haply, divides into 8 frequency separations.That is to say, 500Hz-687.5Hz is made as first interval, 687.5Hz-875Hz is made as second interval, 875Hz-1062.5Hz and is made as the 3rd interval, 1062.5Hz-1250Hz and is made as the 4th interval, 1250Hz-1437.5Hz and is made as the 5th interval, 1437.5Hz-1625Hz and is made as that the 6th interval, 1625Hz-1812.5Hz are made as between SECTOR-SEVEN, 1812.5Hz-2000Hz is made as between Section Eight, frequency signal Fs is carried out 8 coding.The scope of each frequency separation is identical in the present embodiment, but also can set different scopes separately, and the present invention is not as limit.In addition, though present embodiment is done 8 coding with frequency, also can higher resolution divide more sampling range, for example 24,32 or 128 or the like are to obtain finer and smoother frequency change.
Then, shown in step 208, and please refer to Fig. 6, its expression wait to take a sample frequency signal synoptic diagram of sound.Processing unit 130 is according to the frequency sample interval, and the sampling frequency signal SF that treats that treats sampling sound takes a sample.The sound that under noisy environments, sends of user for example.That is to say that processing unit 130 is treated the pairing part for the treatment of that sampling frequency signal SF is positioned at the frequency sample interval of frequency of sampling sound and taken a sample.Because the frequency sample interval is between 500Hz-2000Hz, therefore can not take a sample, and obtain the sound of a frequency of speaking near former user less than the low frequency part of 500Hz or greater than the frequency signal of the HFS of 2000Hz.
In addition, present embodiment can comprise that also treating the pairing sampling frequency signal SF that treats of sampling sound with bandpass filter (bandpass filter) carries out filtering, to filter out most noise in advance between step 204 and step 205.The frequency range of bandpass filter can be set between 100Hz~4000Hz, just the frequency range of common people's sounding.
In the present embodiment, step 201-204 partly does sampling for volume, and step 205-208 then partly does sampling for frequency.In fact, processing unit 130 can be taken a sample to volume and frequency at the same time or separately, therefore also can carry out step 205-208 earlier, carries out step 201-204 again, and the present invention is not as limit.
Embodiment two
Please refer to Fig. 7, the sound signal processing process flow diagram of its expression embodiment of the invention two, and please also refer to the assembly label of Fig. 1, all the other assembly and steps identical with embodiment one illustrate, repeat no more.At first, shown in step 301, and please refer to Fig. 3, the volume according to default sound obtains voltage signal Vs.
Then, shown in step 302, processing unit 130 is according to minimum voltage peak L P and n voltage range of maximum voltage peak value HP definition of voltage signal Vs.The same 8 voltage ranges (n=8) that between 0V-5V, define of present embodiment.Wherein the scope of each 8 voltage range is identical, but not as limit.
Then, shown in step 303, processing unit 130 is set at the volume sampling range with s1 to the t1 voltage range of n voltage range, wherein 1<s1<t1<n.Make n=8 in the present embodiment, and hypothesis minimum voltage peak L V falls between second interval, maximum voltage peak value HV falls between SECTOR-SEVEN between the 3.75V-4.375V.Therefore, can make s1=2, t1=7, setting 0.625V-4.375V is the volume sampling range.
Then, shown in step 304, and,, treat sampling sound and take a sample according to the volume sampling range with reference to figure 4.Just treating the pairing part for the treatment of that sampling voltage signal SV is positioned at the volume sampling range of volume of sampling sound takes a sample.
Then, shown in step 305, and with reference to Fig. 5.Frequency according to default sound obtains frequency signal Fs.
Then, shown in step 306, processing unit 130 is according to minimum frequency peak L F and m frequency separation of maximum frequency peak value HF definition of frequency signal Fs.Present embodiment the same between 500Hz-2000Hz the definition 8 frequency separations.Wherein the scope of 8 frequency separations is identical, but not as limit.
Then, shown in step 307, processing unit 130 is set at the frequency sample interval with s2 to the t2 frequency separation of m frequency separation, wherein 1<s2<t2<m.Make m=8 in the present embodiment, and hypothesis minimum frequency peak L F falls between the 3rd interval 875Hz-1062.5Hz, maximum frequency peak value HF falls between SECTOR-SEVEN between the 1625Hz-1812.5Hz.Therefore, can make s2=3, t2=7, setting 875Hz-1812.5Hz is the frequency sample interval.
Then, shown in step 308, and with reference to Fig. 6.According to the frequency sample interval, treat sampling sound and take a sample.Just treating the pairing part for the treatment of that sampling frequency signal SF is positioned at the frequency sample interval of frequency of sampling sound takes a sample.
In addition, between step 304 and step 305, comprise that also treating the pairing sampling frequency signal SF that treats of sampling sound with bandpass filter carries out filtering.The frequency range of bandpass filter can be set between 100Hz~4000Hz.
In the present embodiment, step 301-304 partly does sampling for volume, and step 305-308 then partly does sampling for frequency.In fact, processing unit 130 can be taken a sample to volume and frequency at the same time or separately, therefore also can carry out step 305-308 earlier, carries out step 301-304 again, and the present invention is not as limit.
Present embodiment allows user's volume and frequecy characteristic be ingested more accurately by setting more accurate sampling scope, keeps the main sound mark of user.
The sound signal processing method that the above embodiment of the present invention is disclosed can be made suitable sampling scope according to user's sound mark, filters out environmental noise.Owing to can adjust the sampling scope according to different users, also can use this function filtering environmental noise for user with special sound mark.
In sum, though the present invention discloses as above with preferred embodiment, so it is not in order to limit the present invention.The technician of the technical field of the invention, without departing from the spirit and scope of the present invention, when doing various changes and improvement.Therefore, protection scope of the present invention is as the criterion when looking the claim person of defining.
Claims (16)
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CN2006101722547A CN101211563B (en) | 2006-12-30 | 2006-12-30 | sound signal processing method |
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CN2006101722547A CN101211563B (en) | 2006-12-30 | 2006-12-30 | sound signal processing method |
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CN101211563B true CN101211563B (en) | 2011-12-21 |
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CN107331404A (en) * | 2017-06-22 | 2017-11-07 | 深圳传音通讯有限公司 | The sound processing method and device of audio frequency and video |
Citations (5)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
US4405948A (en) * | 1982-01-26 | 1983-09-20 | Rca Corporation | Volume control signal coupling circuit in an audio signal processing system |
CN1220056A (en) * | 1996-04-04 | 1999-06-16 | 艾利森公司 | Method for automatically regulating audio response for improved intelligibility |
CN1240316A (en) * | 1998-03-30 | 2000-01-05 | 日本电气株式会社 | Portable terminal equipment for controlling receiving/transmitting phonetic electric level |
CN1299092A (en) * | 1999-12-03 | 2001-06-13 | 张发勋 | Use voice to turn on and off the computer power system |
CN1614883A (en) * | 2004-12-02 | 2005-05-11 | 上海交通大学 | Automatic volume amplitude limiter with automatic gain controlling function |
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- 2006-12-30 CN CN2006101722547A patent/CN101211563B/en not_active Expired - Fee Related
Patent Citations (5)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
US4405948A (en) * | 1982-01-26 | 1983-09-20 | Rca Corporation | Volume control signal coupling circuit in an audio signal processing system |
CN1220056A (en) * | 1996-04-04 | 1999-06-16 | 艾利森公司 | Method for automatically regulating audio response for improved intelligibility |
CN1240316A (en) * | 1998-03-30 | 2000-01-05 | 日本电气株式会社 | Portable terminal equipment for controlling receiving/transmitting phonetic electric level |
CN1299092A (en) * | 1999-12-03 | 2001-06-13 | 张发勋 | Use voice to turn on and off the computer power system |
CN1614883A (en) * | 2004-12-02 | 2005-05-11 | 上海交通大学 | Automatic volume amplitude limiter with automatic gain controlling function |
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