Sampling Rate sampling bit sampling rate: the number of sample samples collected in one second. The reciprocal of the sampling frequency in Hz is the sampling period or the sampling time, which is the time interval between sampling ~ 20 k is the sampling rate of the human's Auditory range 44.1k. That's enough. sampling is to use discrete signals to represent continuous signal nequest frequencies (nycraftsmanship sampling theorem) the sampling frequency is twice the audio signal frequency. before sampling, some analog signals of high frequencies (which cannot be heard by humans) must be filtered out by the low-pass rate device, form a series of discrete signals-three sources of impulsive digital signal noise. The first is the inherent noise produced by components. Almost all components in the circuit generate certain noise during operation, transistor, resistor, capacitor, this noise is continuous, basically fixed and remains unchanged, and the spectrum is widely distributed. In addition to improving the material and production process of components, this noise, there is almost no way to eliminate the second type of noise from the design error of the circuit itself or the defects in the installation process. circuit design errors often lead to slight self-excitation of the circuit. The third type of noise is very extensive, it is also a frequent disturbance noise. for example, Line crosstalk noise transmission noise jitter dither jitter is used to reduce the correlation between quantization noise and input audio signals. The noise signal jitter can reduce the malformed signal, but it will increase the noise in the audio signal. However, the advantage is greater than the disadvantage. The patience of the noise is greater than the patience of the sound distortion. The complex waveform is almost all. the sound is the result of simultaneous waveforms of different frequencies. A sound waveform may contain frequency components of 100,200 and 300Hz with different amplitude. A waveform with multiple frequency components is called a complex waveform. In the real world, sound is not composed of a certain frequency or several frequencies, but is formed by the superposition of many sine waves with different frequencies and different amplitude. Harmonic pitch: the same frequency as the final complex waveform is called pitch frequency or fundamental frequency. The fundamental frequency is always the lowest frequency component in a complex waveform. Audio Splitting: The frequency component higher than the fundamental frequency is called audio splitting. Harmonic: When the audio splitting frequency is an integer multiple of the pitch frequency, these frequencies are called harmonic. A wildcard refers to a spectrum component other than a fundamental frequency. The audio splitting is not necessarily a harmonic. A sound splitting can be a non-harmonic frequency, it can be a sine wave component in the sound that is higher than the fundamental frequency. harmonic distortion total harmonic distortion refers to the extra harmonic composition of the output signal caused by nonlinear elements when the audio signal source passes through the power amplifier. Harmonic distortion is caused by incomplete linearity of the system. The main noise source in the digital filter is the circuit noise introduced by the analog circuit before the digital system and the analog-to-digital conversion process at the input end of the digital system. generate quantitative noise. These noises may be amplified in the calculation of the digital system. Therefore, an appropriate structure is required when designing the digital filter to reduce the impact of input noise on the system performance. The advantages of the IIR filter and the FIR filter are that the design can directly use the results of the simulation filter design, because the simulation filter itself is an infinite long impulse response. Generally, the design process of the IIR filter is as follows: Firstly, the corresponding analog filter (such as the barworth filter and the cherbihof filter) is designed according to the filter parameter requirements ), then, the analog filter is transformed into a digital filter by ing (such as the impulse response invariant method and bilinear ing) to determine the parameters of the IIR filter. The major disadvantage of the IIR filter is that its stability cannot be guaranteed due to feedback. In addition, the feedback may cause the digital operation of the IIR filter to overflow. The most important advantage of the FIR filter is that the system pole does not exist, and the FIR filter is absolutely stable. The FIR filter also ensures linear phase, which is also important in signal processing. In addition, because no feedback is required, the implementation of the FIR filter is simpler than that of the IIR filter. The disadvantage of the FIR filter is that its performance is inferior to that of the same-order IIR filter. However, due to the rapid development of digital computing hardware, this is no longer a problem. Coupled with the introduction of computer-aided design, the design of the FIR filter is also greatly simplified. Based on the above reasons, the FIR filter has a wider signal-to-noise ratio (Signal to Noise Ratio) signal-to-noise ratio: The power ratio between a signal (meaningful information) than the IIR filter) and the background noise (unwanted signal) signal refers to the electronic signal noise from outside the device that needs to be processed through the device. It refers to the irregular extra signal (or information) that does not exist in the original signal generated after the device), and the signal does not change with the original signal. Signal-to-noise ratio is the normal sound signal played back by the Speaker and the noise signal (Power) when there is no signal) the ratio between the maximum non-distortion sound signal strength produced by the audio source and the noise strength sent at the same time. The audio codec MP3 format MP3 format file is essentially still a waveform file. It is a file produced after the digital waveform audio file is compressed and encoded using MP3. MP3 compression encoding is motion image compression encoding International Standard MPEG-1 included in the audio signal compression encoding scheme of the 3rd layer. Different from the general audio compression encoding scheme, MP3 is mainly based on the human auditory psychology and physiological model, some links:
Http://zh.wikipedia.org/wiki/%E6%95%B0%E5%AD%97%E4%BF%A1%E5%8F%B7%E5%A4%84%E7%90%86 Time Domain Frequency Spectrum http://zh.wikipedia.org/wiki/%E9%A2%91%E8%B0%B1http://edu6.teacher.com.cn/tkc507a/index.htmhttp://blog.renren.com/share/56783546/451720614
Electronic Music principle teaching material online tutorial http://pages.uoregon.edu/emi/chinese/index.php? Id = 7
