Mikroişlemci kontrollu adaptif dengeleyici

Abstract

Bu çalışmada, herhangi bir ortamda kurulu bulunan mevcut ses düzenine eklenerek, ortamın akustiğinden veya kullanılan hoparlörlerden kaynaklanan işaret bozulmalarını otomatik olarak giderecek, mikrodenetleyici kontrollü bir dengeleyici tasarlanmış ve gerçeklenmiştir. Yapılan cihazda referans sinyal üreteci olarak bir beyaz gürültü üreteci kullanılmakta olup, bu üreteç, sistemdeki kuvvetlendiricinin girişine bağlanır. Dinlyicilerin bulunduğu noktaya bir mikrofon yerleştirilerek, ortamdaki ses işareti elektriksel işarete dönüştürülür. Daha sonra elde edilen işaret kazancı, otomatik olarak ayarlanan bir kuvvetlendiriciden geçirilir. îstenen seviyeye" getirilen işaret, kazançları +/- 20 dB aralığın da sayısal olarak kontrol edilebilen 10 adet BGF'nin girişlerine uygulanır. Bu filtrelerin çıkışlarından, ortamın hangi frekans bileşenlerini zayıflattığı, hangilerini ise kuvvetlendirdiği belirlenir ve zayıflayan kısımlar kuvvetlendirilir, kuvvetli gelen kısımlar ise zayıflatılır. Son olarakta, filtrelerin çıkışlarından elde edilen işaretler, bir toplayıcı kuvvetlendirici üzerinden sistemin girişine bağlanır. Böylece ses sisteminin girişine uygulanan işaret, adaptif dengeleyici yardımıyla, ortamın akustiğinden etkilenmeden, dinleyicilerin bulunduğu noktada elde edilmiş olur.

Nowadays, a lot of sound system can produce very quality and noiseless signals in 20H2 - 20kHz frequency range. But, because of the limited loud speaker's frequency range, only a limited part of the produced signals can be transmitted to the room. In addition, because of the reasons which effect the room's acoustics, (geometry, echo and sound rejection of the room furniture's and loudspeaker's situation in the room v. b. ) noisely and distortioned sound signals are obtained at the listener's place. In this design, a microcontroller controlled adaptive equalizer which connects to the present system was designed and realized. The designed equalizer automatically improves the transmitted sound signal and listeners, listen more quality and noiseless music sound. Until todays, in the measurement systems, an equalizer and a real-time analyzer was being used for improving the Hifi device's sound quality or linearizing the room's acoustics. An equalizer which is used in the manual system is an adjusment device which increases or decreases the gain of frequency components in the transmitted sound signal. Namely, an equalizer is electronically linearize the room's acoustics. But a real-time analyzer is a measurement device which shows the which frequencies strongly or weakly received. In the measurement system, when adjustment and measurement operations are realizing, real-time analyzer is put the listener's place. A white noise generator which is used as a referance signal source and an equalizer which is adjusted manualy are connected to the input of the system's power amplifier. After this, -v - white noise signal is transmitted to the room from the loudspeakers of the system. Real-time analyzer which has a microphone receive the transmitted sound signal and using this microphone, received sound signal is converted to the electrical signal. Than, obtained electrical signal is amplified by the amplifier of real-time analyzer and this signal is seperated to frequency components by the band pass filters. This signals which are obtained from the band pass filter's outputs show the levels of frequency components. In addition these levels are indicated on the indicators of real-time analyzer. In this way strongly or weakly received frequency components are obtained by the real-time analyzer. Than after, using equalizer, gain of weakly received frequency components are increased and gain of strongly received frequency components are decreased. At the final, a flat frequency characteristic or white noise is again obtained at the listener's place. An adaptive equalizer which is designed in this study is an automatic measurement and correction device which connects the real-time analyzer and equalizer. But in the adaptive equalizer, measurement and correction procedure of the sound signal are automatically realized. In the measurement and correction system which realized with the adaptive equalizer, a white noise generator is used as a referance signal source. In the adjustment procedure, the white noise generator is connected to the system's power amplifier input and white noise is transmitted to the room with the loudspeakers. Because of the room's acoustics, level of some frequency components in the white noise increase and some of them decrease. This distortioned white noise is received with a microphone which is placed at the listener's place. Received sound signal is converted to the electrical signal by this microphone. Than, distortioned signal level is increased for the level of need to be adjustment procedure by an input amplifier. The gain of this input amplifier is positive and digitally controlled by the system's microcontroller. Output of the input amplifier is connected to the inputs of 10 Band Pass Filters. These filters are -vi- second order, Butterworth type, band pass filters. Their center frequencies are 30Hz, 60Hz, 12SHz, 250Hz, SOOHz, lkHz, 2kHz, 4JcHz, 8kHz, 16kHz and value of filter's Q factor is equal to 1.23. These band pass filters are used for seperate the frequency components of the distortioned signal. Outputs of these filters are connected to the digitally controlled amplifiers. These amplifiers are increase or decrease the band pass filter's gain and levels of the frequency components is adjusted by these amplifiers. The gain of these amplifiers can be controlled in the +\- 20 dB range. These amplifier's outputs are connected to the an adder amplifier which has ten input. This adder amplifier's gain is one and levels adjusted frequency components are added in this amplifier. The adder amplifier's output is the adaptive equalizer's output and the original sound is obtained at this output. Output of the adder amplifier or namely, output of the adaptive equalizer is connected to the inputs of 10 band pass filters. The center frequencies of these filters are same as the other band pass filters. These filters are used to seperate the frequency components of original signal which obtained from adaptive equalizer's output. Outputs of 10 band pass filters are connected to the 10 active half-wave rectifier and 10 passive low pass filters. 10 rectifiers and filters are produce DC voltage which is proportional- to the levels of frequency components. These DC voltages show to the microcontroller, which frequencies low level received and which frequencies high level received. Than, these analog DC voltages is converted to the digital signal by an 8 bit analog to digital converter and 10 to 1 analog demultiplexer system. 10 to 1 analog demultiplexer system is used for reduce the number of analog to digital converter. This demultiplexer system and analog to digital converter are controlled by the system's microcontroller. At the last, microcontroller looks the digital signals and then increases the gain of weakly received frequencies and decreases the gain of strongly received frequencies. In this way at the listeners place, the original sound signal is obtained by the microcontroller controlled adaptive equalizer. -vii- In this study, 80C51BH CMOS microcontroller is used for controlling the band pass filters gain. This microcontroller is a member of Intel MCS51 " 8 bit microcontroller family. This CMOS version require less power than the NMOS version and it's operating speed is higher than the NMOS version. Instruction tables are same with each other but pin configurations are different from each other. The features of CMOS microcontrollers are same as the NMOS microcontrollers and the major 8051 Family features are: * 8-Bit CPU optimized for control applications * 4K bytes of on-chip Program Memory * 128 Bytes of on-chip Data Memory * Two 16-bit Timer /Counter s * Full duplex UART ** S-source interrupt structure with two priority levels * On-chip oscillator * Boolean Processor * Bit-addressable RAM * 64K Program Memory Space * 64K Data Memory Space The 80C51BH has two different power saving mod, idle and power -down mod. In idle mod, the CPU is turned off while the RAM and other on-chip peripherals continue to operate. In this mode, current draw is typically 1S?« of fully active current draw. In the power -down mode, all on-chip activities are suspended while RAM holds its data. In this mode, current draw is typically less than 10 uA. The 80C51BH microcontroller has seperate address spaces for program memory and data memory. The Program Memory can be up to 64K bytes long. The Data Memory can consist of up to 64K bytes of off -chip- RAM, in addition to which it includes 128 bytes of on-chip RAM, plus a number of Special Function Registers. C Accumulator, Program Status Word, Stack Pointer, Data Pointer, Power Control etc. 5. The 80C31BH and 87CS1 are also member of the 8051 family. But 80C31BH contains no on-chip ROM and the 87CS1 is the EPROM version of the 80CS1BH. -viii In this design, using instruction table of 80C51BH microcontroller, an assembler program was written for controlling the band pass filters gain. This program was developed on the EV80C51 evaluation, board. EV80C51 evaluation board is a design equipment which is used for developing software and hardware. of 80C51 microcontroller. This evaluation board communicate with an IBM PC XT/AT via the RS332 serial interface. When the software developments are made on the EV80C51 evaluation board, the assembler program was written on the IBM PC editor. Then, this program is compiled and converted to the instruction codes of 80C51BH. After the compiling procedure, this codes are transmitted to the evaluation board by using the IRISM51 program. The IRISM51 is a software program which is prepared for providing the communications of the IBM PC XT/AT and evaluation board.