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1. Software-defined radio is a solution to the coexistence of multiple systems in the field of radiocommunications and the inability to formulate a unified standard between different systems. Becausesoftware radio implements various functions based on software programming, its main characteristicsare manifested in flexibility and openness. Software-defined radio is mainly a communication softwareto rely on. This software can complete functions such as wireless calls and video surveillance withsoftware programming. It has features such as functionalization and modularity. As long as the signalfrequency band that its hardware system can process, the communication function in the correspondingfrequency band can be increased by software. Digital signal processing technology has thecharacteristics of stable signal, fast transmission and strong anti-interference. It can be seamlesslyconnected with software radio and promote the application and development of software radio.Software radio is a multi-band radio. It has a wideband antenna, a radio frequency front end, and ananalog-to-digital / digital-to-analog conversion. It can support multiple air interfaces and protocols. Inan ideal state, all aspects (including the physical air interface) can pass Software to define. The idealsoftware radio uses wideband analog-to-digital converters and digital-to-analog converters to performA / D, D / A conversion at the antenna port, and digital signal processing at the back end is implementedby software of programmable devices. The structure of the radio is shown in Figure 4.It is a characteristic of software-defined radio architecture to place the analog-to-digital converterand digital-to-analog converter as close to the RF front-end as possible. The status of the analog-todigital converter and digital-to-analog converter in software-defined radio systems is also critical, whichwill directly affect the performance indicators of the entire system are excellent. At present, the hardware implementation of digital processing mainly depends on two types of highspeed DSP chips and FPGAs. The respective characteristics of DSP and FPGA are relatively obvious,which are suitable for different occasions. At present, a development direction that is relatively wellrecognized by everyone is the structure of DSP and FPGA. The purpose is to make the most of theadvantages of these two processors and complement each other to achieve the parallel processing ofsignals in the entire system. When the clock is limited, the processing power of the system is improvedas much as possible.
2. The main purpose of speech compression coding is to obtain clear and high-quality speech throughcorresponding equipment to convert information. Therefore, its transmission signal must have stronganti-interference ability and be able to transmit information in a narrow bandwidth spectrum in order toachieve reception. And fully restore the transmitted voice information to facilitate informationcommunication. At the beginning of the speech coding system, the coding method mainly adopted largewaveform coding, fully following the sampling guidelines of discrete Fourier transform, focusing on theuse with the external environment, and ensuring the high-quality state of the speech signal. It is relativelyfast, and it is easy to appear a series of problems in use, which affects the final voice signal quality.Another common form of speech coding is parameter coding. The difference is that the key parametersof the speech signal are used as the coding, but the coding rate is relatively low, which affects the speecheffect.A typical compression-sensing-based speech compression coding and reconstruction framework isshown in Figures 2 and 3. In a speech compression system, it is mainly composed of a speech encoder, a digital storage medium,and a decoder. The speech input is dominated by the speech encoder, and the speech output is led by thespeech decoder. The information transmission process is mainly the speech input of the speech encoder? the digital storage medium ? the speech output of the speech decoder, which is the simplified speechcompression coding. With the development of the times, the voice compression system mainly consistsof a voice input module, a DSP module, an A / D conversion module, a D / A conversion module, anda voice output module. The key technology is the DSP module, which is the voice. Compression anddecompression need to be implemented with the help of special algorithms and techniques to ensure thatthe voice signal is not damaged. The DSP chip has simple data processing characteristics and canimplement multiplication and addition operations in a single instruction cycle. It is more suitable forvoice compression systems. In addition, various hybrid encodings have also gradually emerged. Forexample, linear predictive encoding, as a new type of speech encoding technology, has a processingspeed of 4 to 16 kilobytes per second, but it needs to match higher algorithms and digital signalprocessors. The digital signal processor can solve these problems, effectively improve the problem ofvoice transmission, and improve the stability and reliability of the voice compression system.