0 Preface
With the development of society and people's increasing attention to hearing-impaired patients, the development of hearing aids has gradually received attention. However, due to the different causes of hearing-impaired patients, there are large differences in hearing loss, which makes each patient have different requirements for the compensation of hearing aids. At present, modern hearing aid technology has entered the era of all-digital hearing aids. At the same time, various digital signal processing algorithms that effectively improve the performance of digital hearing aids have received more attention. The digital hearing aid design based on TMS320VC5416 is proposed to meet the hearing needs of hearing-impaired patients.
l System structure and working principle
1.1 System components
Based on the technical requirements of the hearing aid, TI's C54X series products TMS320C5416 (hereinafter referred to as C5416) and digital encoder TLV320AIC23 (hereinafter referred to as AIC23) are selected.
The digital encoder AIC23 is a high-performance stereo audio Codec chip introduced by TI. The A/D conversion and D/A conversion components are integrated inside the chip, using advanced ∑-Δ oversampling technology and built-in headphone output amplifier. The AIC23DSPCodec's operating voltage is compatible with the C5416's core and I/O voltages, enabling seamless connection to the C54x serial port and low power consumption, making the AIC23 an ideal audio analog device for digital applications. The design of the hearing aid.
The system structure is shown in Figure 1. It mainly includes DSP module, audio processing module, JTAG interface, storage module and power module. The analog voice signal is input to the AIC-23 through the MIC or IANE IN. After the analog/digital conversion, the C5416 is input through the MCBSP serial port. After the actual required algorithm is processed and compensated, the voice signal required by the hearing impaired patient is obtained, and then the AIC23 is passed. Digital/analog conversion, outputting sound signals through speakers or headphones.
1.2 Interface design of C5416 and AIC23
Figure 2 is a schematic diagram of the interface between C5416 and AIC23. Since the AIC23 samples the serial data, it is necessary to coordinate the serial transmission protocol of the matched DSP. The MCBSP is most suitable for voice signal transmission. Connect the 22nd pin of the AIC23 to the high level and receive the serial data of the SPI format from the DSP. The digital control interface (SCLK, SDIN, CS) is connected to the MCBSP1. The control word has a total of 16 bits and is transmitted from the high bit. The digital audio ports LRCOUT, LRCIN, DOUT, DIN, BCLK are connected to MCBSP0. In the working mode, DSP is the main mode, and AIC23 is the slave mode, that is, the clock signal of BCLK is generated by the DSP.
The serial clock is connected by BCLKX0 and BCLKR0 in parallel to the BCLK clock of AIC23, so that the serial clock signal can be generated when transmitting and receiving data. The input/output sync signals LRCIN and LRCOUT are used to initiate serial port data transmission and receive the frame sync signal of the DSP.
BFSX0 and BFSR0, BDR0 and BDX0 are connected to the DIN and DOUT of AIC23 respectively to implement digital communication between DSP and AIC23.
2 system implementation
2.1 Basic characteristics of speech
The sound is a kind of wave, and the vibration frequency that can be heard by the human ear is 20 Hz to 20 kHz. Speech is a kind of sound. It is a sound made by human vocal organs with certain grammar and meaning. The voice has a vibration frequency of up to 15 kHz.
Voice is divided into different types of motivation: voiced, unvoiced, and plosive. The human voice characteristics are basically determined by factors such as the gene cycle and formants. When a voiced sound is heard, the airflow vibrates the vocal cords through the glottis, producing a quasi-periodic excitation pulse train. The period of this burst is called the "gene cycle" and the reciprocal is the "gene frequency."
Both the human vocal tract and the nasal passage can be regarded as a channel tube with a non-uniform interface. The resonant frequency of the channel tube is called a formant. Changing the shape of the channel produces a different sound. The formant is represented by multiple frequencies that increase in sequence. Such as F1, F2, F3, etc., referred to as the first formant, the second formant, and the like. In order to improve the quality of speech reception, as many formants as possible must be used. In practice, the first three formants are the most important, and the specific situation varies from person to person.
2.2 Voice Enhancement
In the actual application environment, the voice will be disturbed by environmental noise to varying degrees. Speech enhancement is the processing of noisy speech, reducing the effects of noise and improving the auditory environment.
The interference encountered by actual speech may include the following categories:
(1) Periodic noise: such as electrical interference, interference caused by engine rotation, etc., such interference appears as discrete narrow peaks in the frequency domain. In particular, 50 Hz or 60 Hz hum can cause periodic noise.
(2) Impact noise: such as electric sparks, noise interference caused by discharge, such interference appears in the time domain as a sudden burst of narrow pulses. Eliminating this noise can be done in the time domain by determining the threshold based on the average of the amplitude of the noisy speech signal.
(3) Wideband noise: usually refers to Gaussian noise or white noise, which is characterized by a frequency bandwidth that covers almost the entire voice frequency band. It has many sources, including wind, respiratory noise, and general sources of random noise.
2.3 Algorithm Analysis
The influence of noise makes the language recognition rate of the patient drop sharply. Denoising and compensation are important parts of the hearing aid. The human ear responds to sounds from 25 to 22 000 Hz. Most of the available information for speech exists only between 200 and 3 500 Hz. According to the human ear perception characteristics and experimental determination, the second resonance peak, which is important for speech perception and speech recognition, is mostly above 1 kHz.
2.3.1 Periodic noise cancellation
Periodic noise is generally a number of discrete spectral peaks derived from the periodic operation of the engine. Electrical interference, especially 50 to 60 Hz hum, can also cause periodic noise. Therefore, the use of a bandpass filter can effectively eliminate periodic noise and high frequency sound above 3 500 Hz.
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