Design Of Wireless Voice Transmission Control System

With the rapid development of electronic technology,the innovation of wireless communication network technology has promoted the advancement and convenience of multimedia equipment.For the expanding environment of living environment and working environment,the effective transmission of audio and the facility and convenience of devices become the new pursuit of voice devices.Speech transmission and implementation in large conference rooms,cars and other fields,complex cabling space and other environments,it has become a research hotspot on the basis of humanized and convenient devices to achieve efficient speech quality transmission.The wireless voice transmission control system is designed and developed based on these factors and requirements,which greatly improves the shortcomings of voice transmission equipment,and is difficult to install and route.The overall design of the system can be composed of two parts: hardware and software.The hardware mainly includes audio acquisition and receiving module design,the design of wireless transmission module,voice playing module,the main control with low power,high performance ARM series processor STM32F105,RF transceiver chip n RF24L01 as a wireless platform,the design process,focusing on the amplification of the audio signal acquisition and processing and filtering.The development of control system software mainly includes the compilation of embedded software,namely speech signal processing,audio signal processing algorithm,communication algorithm of n RF24L01 chip,and control program design of control system peripherals.The design is mainly used for C++ language programming,in order to simplify the principle,to write a strong readable and high stability control program.In the aspect of algorithm realization,the ADPCM(adaptive differential pulse code modulation)algorithm is optimized and improved.The correlation between the signal sampling value,and combined with the characteristics of non-stationary speech signal,the implementation of adaptive prediction and adaptive quantization is used to denoise the speech signal at the same time,using the LMS adaptive filter in noise reduction algorithm,which greatly improves the quality of voice transmission system.Building a test platform,debugging and analysis of system function,list calculation of the important experimental parameters,analyze the oscilloscope tracking of the important link of the system working process.Finally,the experimental results show that the average transmission accuracy of speech transmission control system can reach 97.32%,meet the performance requirements of the system design.