CN104902419B - Frequency shift compression method suitable for digital hearing aid - Google Patents
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Abstract
The invention provides a frequency shift compression method suitable for a digital hearing aid, which comprises the following steps: s101, determining the high-frequency loss degree of a patient, and obtaining a frequency range needing to be compressed and a target frequency range; s103, preprocessing the voice; s105, calculating a compression ratio p according to the frequency range needing to be compressed and the target frequency range, and compressing the frequency signal at a compression ratio (p + 1); s107, carrying out symmetrical processing, and carrying out N-point IFFT; and S109, overlapping and adding the converted voice and the signal of the previous frame, and performing automatic gain compensation on the output signal. The frequency shift compression method suitable for the digital hearing aid not only reserves the voice information of the target frequency band, but also can compress the high-frequency band to be compressed, thereby better improving the recognition degree and intelligibility of the voice and simultaneously compensating the voice energy loss caused by frequency shift.
Description
Technical field
The present invention relates to Hearing aid technology more particularly to a kind of shift frequency compression methods suitable for digital deaf-aid.
Background technique
Shift frequency compression algorithm is one of most important algorithm in digital deaf-aid, especially to those phonosensitive nerve hearing
Lose patient.Hearing aid mainly amplifies (single or multi channels) to voice, however, for phonosensitive mind
For property hearing loss patient, they are general, and low frequency loss is smaller, but high frequency loss is very high.When loss reaches 60~70dB
Or when above, if high frequency is compensated, is that it is impossible to meet demands.Itself the reason is as follows that: (1) it is general at present
The gain of hearing aid Amplitude amplification and frequency bandwidth are limited by microphone and speaker performance, and compensating gain generally can only achieve
60db or so.(2) for patient when high frequency treatment loss reaches 60dB or more, the range of audibility is very narrow, is as shown in Figure 1 a patient
Characteristic spectra the threshold of audibility and threshold of pain figure, it is found that patient in 4k or less frequency range there are about the range of audibility of 40~75db, and it is high
Frequency only has the range of 15~30db.(3) related to the sound physical properties characteristic of consonant in voice, most center frequencies in voiceless consonant
Rate is located at 4khz or more, since hearing aid is extremely difficult to the gain of 20dB or more in high frequency, and has 35% important voice to concentrate
In 2khz or more, it is difficult to improve patient to lalognogis only by amplification.(4) diffusion mask characteristic acoustically, this spy
Property determine that low-frequency sound has more shelterings to high frequency sound, due to low frequency amplification and high frequency is put less, cause vowel to auxiliary
The sheltering of sound, speech discrimination decline.
But human ear is not but the relative ratios of frequency by absolute frequency to the resolution capability of sound, for that
A little very grave degree phonosensitive nerve patients, although high frequency loss is serious, the residual hearing range of its low frequency or bigger.It is existing
Having feasible method is usually that High frequency speech is compressed at low frequency, to improve the audible frequency range of patient.
The research of early stage mainly simple carry out frequency compression, equal proportion and nonlinear way, so but largely
On change the clarity and intelligibility of voice, while the speech energy after compressing changes significantly, also affects patient to language
The identification of sound.Based on the considerations of this two o'clock, relationship of the present invention according to shared by different frequency range voice between energy and intelligibility is mentioned
A kind of shift frequency compression method suitable for digital deaf-aid based on the conservation of energy out, can preferably solve the above problems.
Summary of the invention
Based on this, it is necessary in view of the above-mentioned problems existing in the prior art, provide a kind of shifting suitable for digital deaf-aid
Frequency compression method, it is of the existing technology to solve the problems, such as.
A kind of shift frequency compression method suitable for digital deaf-aid comprising following steps:
S101, the high frequency loss degree for determining patient, acquisition need compression frequency range and range of target frequencies;
S103, voice is pre-processed;
S105, compression frequency range and range of target frequencies are needed according to described, compression ratio p is calculated, with compression ratio (p+1)
Carry out frequency signal compression;
S107, symmetrical treatment carry out N point IFFT transformation;
S109, the voice switched back to are added with previous frame signal overlap, and the signal of output carries out automatic gain compensation.
In a better embodiment of the invention, in step S103, the sample frequency of voice is 16khz, every frame 16ms, frame move
It is N for 8ms, FFT size, before fft processing, carries out hamming windowing process.
In a better embodiment of the invention, in step S109, the average sound pressure level of L frame before calculating first, further according to defeated
Voice sound pressure level out.
It further comprise that the voice sound pressure level of output is handled as follows: first in a better embodiment of the invention
The average gain value gain_pre of L frame, then uses formula before countingMeter
It calculates, the sound pressure level compensating gain as present frame output.
It is as follows to the update of gain_pre in a better embodiment of the invention:
Wherein, gain_o is voice sound pressure level after compensation
Compensating gain, gain_out be it is smooth after compensating gain.
In a better embodiment of the invention, in step S109, when SPL_ori is less than the threshold of audibility of patient, then without carrying out
Energy compensating, it is believed that be not voice;When the threshold of audibility of the SPL_ori greater than patient, then energy compensating is carried out.
Compared to the prior art, the shift frequency compression method provided by the invention suitable for digital deaf-aid both remains target
The voice messaging of frequency range, and the high-frequency band compressed can will be needed to compress, preferably improve the identification of voice with
Intelligibility, while also compensating for the loss of the speech energy as caused by shift frequency.
Detailed description of the invention
Fig. 1 is the threshold of audibility and threshold of pain figure of the characteristic spectra of a patient;
Fig. 2 is the flow chart of the shift frequency compression method provided by the invention suitable for digital deaf-aid;
Fig. 3 is fohFor the obtained voice time domain figure of 3khz;
Fig. 4 is the work flow diagram for being suitable for the shift frequency compression method of digital deaf-aid described in one embodiment of the invention;
Fig. 5 be the present invention construct based on the relational graph between frequency and speech energy and its intelligibility;
Fig. 6 is primitive sound by conventional method and using the shift frequency pressure for being suitable for digital deaf-aid provided by the invention
The comparison diagram of contracting method sound result obtained;
Fig. 7 is the spectrum pair of conventional method and the shift frequency compression method suitable for digital deaf-aid provided by the invention
Than figure.
Specific embodiment
To facilitate the understanding of the present invention, a more comprehensive description of the invention is given in the following sections with reference to the relevant attached drawings.In attached drawing
Give better embodiment of the invention.But the invention can be realized in many different forms, however it is not limited to herein
Described embodiment.On the contrary, the purpose of providing these embodiments is that making to understand more the disclosure
Add thorough and comprehensive.
It should be noted that it can directly on the other element when element is referred to as " being fixed on " another element
Or there may also be elements placed in the middle.When an element is considered as " connection " another element, it, which can be, is directly connected to
To another element or it may be simultaneously present centering elements.Term as used herein " vertical ", " horizontal ", " left side ",
" right side " and similar statement for illustrative purposes only, are not meant to be the only embodiment.
Unless otherwise defined, all technical and scientific terms used herein and belong to technical field of the invention
The normally understood meaning of technical staff is identical.Term as used herein in the specification of the present invention is intended merely to description tool
The purpose of the embodiment of body, it is not intended that in the limitation present invention.Term " and or " used herein includes one or more
Any and all combinations of relevant listed item.
Referring to Fig. 1, the present invention, which provides the present invention, provides a kind of shift frequency compression method suitable for digital deaf-aid, wrap
Include following steps: S101, the high frequency loss degree for determining patient, acquisition need compression frequency range and range of target frequencies;
S103, voice is pre-processed;S105, compression frequency range and range of target frequencies are needed according to described, calculates compression ratio
P carries out frequency signal compression with compression ratio (p+1);S107, symmetrical treatment carry out N point IFFT transformation;S109, switch back to come
Voice is added with previous frame signal overlap, and the signal of output carries out automatic gain compensation.
Studies have shown that the intelligibility at the frequency of voice signal and speech energy and the frequency has pass as shown in Table 1
System.
Table 1: the relationship of frequency and speech energy and intelligibility
Frequency (Hz) | Speech energy (%) | Intelligibility (%) |
60~125 | 5 | 1 |
125~250 | 13 | 1 |
250~500 | 42 | 3 |
500~1000 | 35 | 35 |
1000~2000 | 3 | 35 |
2000~4000 | 1 | 13 |
4000~8000 | 1 | 12 |
From table 1 it follows that the energy of voice is concentrated mainly on the low frequency part of 125hz~2000hz, 93% left side is accounted for
It is right;And the intelligibility of speech of the high frequency section of 1khz~8khz accounts for the high frequency section of 60% or more, especially 2khz~8khz, language
Sound energy only accounts for 2% or so, and intelligibility has but reached 25%, so the voice of high frequency section has very big work to speech understanding
With wherein 500hz~2000hz voice is particularly important.According to another perhaps big et al. research (Xu Wei, Zeng Xinwu, Gong Changchao, different band
Wide and the sample frequency intelligibility of speech the experimental study [A], national acoustics academic meeting paper collection [C] in 2008,2008),
The voice in " electro-acoustic product subjective assessment program source " national standard CD is utilized, different frequency resampling is carried out and is tested
Source, by the subjective judgement of 8 people's auditions as a result, carrying out region filtering to 0~1khz voice, examination different frequency range missing is to voice
The influence of quality finds that under the requirement that voice content can be understood completely, the minimum upper limit of speech frequency should take in 1khz, most relative superiority or inferiority
Limit is in 300hz;Meanwhile speech sample frequency be higher than 7khz when, voice quality is had not significant impact, sample frequency is in 7khz
The decline of~4khz voice quality is obvious, but can understand completely, and when sample frequency is lower than 4khz, semantic understanding becomes more difficult.
By being researched and analysed above it is found that the voice of 300hz~1khz is especially important, is in carrying out shift frequency compression process
It can not be destroyed, the voice of 1khz~2khz should retain as far as possible.In the present invention, selection for target frequency, one is
Based on patient's high frequency degree of injury, one is based on relationship (inventor's structure between frequency and speech energy and its intelligibility
A kind of conversion relationship built).
Shift frequency compression method provided by the invention suitable for digital deaf-aid is non-linear shift frequency compression method, is improved
Also reside in the compensation to energy.In hearing aid of the invention, under normal circumstances, shift frequency compression method is placed on WDRC (Wide
Dynamic Range compression, wide dynamic range compression) it carries out later to speech processes.After WDRC, according to trouble
The actual conditions of person have carried out certain gain to voice, have been dropped after the algorithm, and gain.Such as Fig. 3 institute
Show, is fohFor the obtained voice time domain figure of 3khz, the amplitude of audio is significantly had dropped, and such shift frequency just affects sound
Loudness.In response to this, the present invention proposes the method that sound pressure level energy compensating is carried out in time domain.
Since for compression ratio difference, energy loss is different, so energy compensating mode, it is desirable to be able to automatic gain
It adjusts, the principle of compensation counts the average energy of current frame speech first:
Again by sound pressure level conversion formula:
Wherein: s_ini(n) be the i-th frame primitive sound signal, XiIt (n) is frequency domain information after i frame voice FFT, pref
It is constant 20upa, N is the length of FFT.
Due to the short-term stationarity of voice, need to count front L frame, with maintain will not be because of the time-domain diagram to voice caused by
Very big gain error and introduce noise.
Seek compensating gain gain formula:
Compensated signal can be acquired.
Referring to Fig. 4, the shift frequency compression method suitable for digital deaf-aid includes as follows in one embodiment of the invention
Step:
S101, the high frequency loss degree for determining patient, acquisition need compression frequency range and range of target frequencies.
Specifically, it is first determined the high frequency loss degree namely cutoff frequency freq of patient;Then according to compression ratio p come
Determination needs compression frequency range and range of target frequencies.The present inventor constructs conversion relationship, that is, is based on by repeatedly measurement
Relationship between frequency and speech energy and its intelligibility, as shown in Figure 5.
S103, voice is pre-processed.
Specifically, voice is pre-processed, the sample frequency for the voice that the present invention uses is 16khz, every frame 16ms, frame
It is N that shifting, which is 8ms, FFT size, before FFT (Fast Fourier Transformation, fast Fourier transform) processing, into
Row hamming windowing process.
S105, compression frequency range and range of target frequencies are needed according to described, compression ratio p is calculated, with compression ratio (p+1)
Carry out frequency signal compression.
Specifically, compression frequency range and range of target frequencies are needed according to what step S101 was determined, calculate compression ratio p,
Frequency signal compression is carried out with compression ratio (p+1).
It is noted that compression ratio is p+1, the signal in target frequency be not it is directly capped, but also to be compressed,
It thus could include prior information.
S107, symmetrical treatment carry out N point IFFT transformation.
Inverse fast Fourier transform is carried out, signal from frequency-domain transform to time domain.
S109, the voice switched back to are added with previous frame signal overlap, and the signal of output carries out automatic gain compensation.
Specifically, switching back to the voice come to be added with previous frame signal overlap, the signal of output carries out automatic gain compensation,
The average sound pressure level of L frame is compensated further according to the voice sound pressure level of output by mode as shown in Figure 5 before calculating first.
When SPL_ori is less than the threshold of audibility of patient, then without carrying out energy compensating, it is considered that be not voice;Work as SPL_ori
Greater than this threshold value, by progress energy compensating shown in figure.
Here in order to be unlikely to gain inequality between every frame voice away from too big, and the short-term stationarity of voice is impacted,
The present embodiment has carried out following processing to the sound pressure level compensating gain of output: the average gain value gain_pre of L before counting first,
Then it is calculated with formula below:
As the sound pressure level compensating gain of present frame output, meanwhile, it is as follows to the update of gain_pre:
Wherein: gain_o is the compensating gain of voice sound pressure level after compensation, gain_out be it is smooth after compensating gain.
Fig. 6 show primitive sound by conventional method and using the shifting for being suitable for digital deaf-aid provided by the invention
The comparison diagram of frequency compression method sound result obtained, can obviously find from figure, be suitable for digital deaf-aid by described
Shift frequency compression method processing after sound intensity it is bigger.
Referring to Fig. 7, for conventional method and the shift frequency compression method suitable for digital deaf-aid provided by the invention
Spectrum comparison diagram, therefrom, can more intuitively find out voice frequency content variation, voice is mainly concentrated before untreated
In 2khz~8khz;After the shift frequency compression method processing suitable for digital deaf-aid, radio-frequency component is seldom, low frequency
Ingredient is reinforced, and former high-frequency signal is strengthened in intermediate frequency zone.
Table 2 is the compensation sound of conventional method and the shift frequency compression method suitable for digital deaf-aid provided by the invention
It arbitrarily downgrades comparison, i.e., the one section of language exported after conventional method and the shift frequency compression method suitable for digital deaf-aid are handled
The comparison of the average sound pressure level of sound.It can be found that cutoff frequency is lower namely compression ratio is bigger, automatic gain energy compensating
Effect is more obvious, and output sound pressure level is higher.
Table 2 compensates sound pressure level comparison
Compared to the prior art, the shift frequency compression method suitable for digital deaf-aid provided by the invention both remained
The voice messaging of target frequency bands, and the high-frequency band compressed can will be needed to compress, preferably improve the identification of voice
Degree and intelligibility, while also compensating for the loss of the speech energy as caused by shift frequency.
The embodiments described above only express several embodiments of the present invention, and the description thereof is more specific and detailed, but simultaneously
Limitations on the scope of the patent of the present invention therefore cannot be interpreted as.It should be pointed out that for those of ordinary skill in the art
For, without departing from the inventive concept of the premise, various modifications and improvements can be made, these belong to guarantor of the invention
Protect range.Therefore, the scope of protection of the patent of the invention shall be subject to the appended claims.
Claims (3)
1. a kind of shift frequency compression method suitable for digital deaf-aid, which comprises the steps of:
S101, the high frequency loss degree for determining patient, acquisition need compression frequency range and range of target frequencies;
S103, voice signal is pre-processed, carries out N point FFT transform;
S105, compression frequency range and range of target frequencies are needed according to described, calculates compression ratio p, with compression ratio (p+1) progress
Frequency-region signal compression;
S107, symmetrical treatment carry out N point IFFT transformation;
S109, the voice signal switched back to and previous frame voice signal overlap-add, the voice signal of output are increased automatically
Benefit compensation,
Wherein in step S109, the sound pressure level of the voice signal of output is handled as follows: L frame is averaged before counting first
Yield value gain_pre, then uses formulaIt calculates, as
The compensating gain of the sound pressure level of the voice signal of present frame output, thus realize automatic gain control and sound pressure level energy compensating,
It is wherein as follows to the update of gain_pre:
Wherein, gain_o is compensated voice letter
Number sound pressure level compensating gain, gain_out be it is smooth after compensating gain.
2. being suitable for the shift frequency compression method of digital deaf-aid as described in claim 1, which is characterized in that in step S103,
The sample frequency of voice signal is 16khz, the shifting of every frame 16ms, frame be 8ms, FFT size is N, before fft processing, is carried out
Hamming windowing process.
3. being suitable for the shift frequency compression method of digital deaf-aid as described in claim 1, which is characterized in that in step S109,
When original sound pressure level SPL_ori is less than the threshold of audibility of patient, then without carrying out sound pressure level energy compensating, it is believed that be not voice signal;
When the threshold of audibility of the original sound pressure level SPL_ori greater than patient, then sound pressure level energy compensating is carried out.
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CN108281148B (en) * | 2016-12-30 | 2020-12-22 | 宏碁股份有限公司 | Speech signal processing apparatus and speech signal processing method |
CN110830897B (en) * | 2018-08-08 | 2021-04-09 | 原相科技股份有限公司 | Hearing aid and method for adjusting output voice of hearing aid |
CN109327785B (en) * | 2018-10-09 | 2020-10-20 | 北京大学 | Hearing aid gain adaptation method and device based on speech audiometry |
CN109862463A (en) * | 2018-12-26 | 2019-06-07 | 广东思派康电子科技有限公司 | Earphone audio playback method, earphone and its computer readable storage medium |
CN111755023B (en) * | 2020-04-15 | 2023-10-13 | 欧仕达听力科技(厦门)有限公司 | Frequency shift real-time loudness compensation method based on equal loudness curve |
CN113362839A (en) * | 2021-06-01 | 2021-09-07 | 平安科技(深圳)有限公司 | Audio data processing method and device, computer equipment and storage medium |
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