Common sense concepts such as sound, audio sampling rate, and Sampling Accuracy

What is sound?

Energy Wave, with frequency and amplitude, the frequency level is the tone, the amplitude size is the volume, the sampling rate is the frequency sampling, and the sampling accuracy is the amplitude sampling

The frequency range that people can hear is 200-20 kHz,

What is codec?

Audio Compression Co + extract Dec, codec is a multimedia digital signal decoder, mainly responsible for DAC and ADC. Whether the audio accelerator is good or the I/O controller is good, their input and output are pure digital signals. If we want to use the line out Jack on the sound card to output signals, the signal must undergo codec conversion on the sound card. It can be said that the analog input and output quality of the sound card has a major relationship with the codec conversion quality. The audio accelerator or I/O Controller determines the quality of the digital signal inside the sound card, codec determines the quality of analog input and output.

What is dac adc?

Modulus, digital-to-analog conversion, different precision corresponds to different sound fidelity

What is EQ?

The value range is-15db-15db.

BTL principles

(Bridge-tied-load) indicates a bridge load.

The two ends of the load are connected to the output ends of the two amplifiers respectively. The output of one amplifier is the image output of another amplifier. That is to say, the signal added at both ends of the load is only 180 ° different in phase. Load will generate 2 times the voltage of the original single-ended output. Theoretically, the output power of the circuit will increase by 4 times. The BTL Circuit can make full use of the system voltage. Therefore, the BTL structure is often used in low-voltage systems or battery power supply systems.

What is SRC?

Sound is an energy wave characterized by frequency and amplitude. The frequency corresponds to the time axis and the amplitude corresponds to the level axis. The wave is infinitely smooth, and the string line actually consists of numerous points. Because the storage space is relatively limited, the point of the string line must be sampled during the digital encoding process. The sampling process is to extract the frequency value of a certain point. Obviously, the more points are obtained in one second, the more frequency information is obtained. In order to restore the waveform, in one vibration, there must be two sampling points, and the highest frequency that the ears can feel is 20 kHz. Therefore, to meet the auditory requirements of the ears, at least 40 kb of sampling per second is required and expressed at 40 kHz, this 40 kHz is the sampling rate. Our common CD, sampling rate is 44.1 kHz. It is not enough to have frequency information. We also need to obtain and quantify the energy of the frequency to indicate the signal strength. The Quantization value is an integer power of 2. We usually use a 16-bit CD sample size, that is, the 16 power of 2.

The function of SRC is to change the sampling rate of the signal. The conversion from low sampling rate to high sampling rate is a re-sampling process. The re-sampling object is not the original signal, but the signal at low sampling rate, because the sampling rate is not enough, more sampling points need to be inserted to achieve the required sampling rate and sampling size. When the signal frequency is low, the quality of the re-sampling algorithm will not affect anything, because the wavelength is long, there are many sampling points, but the high frequency is difficult to deal with, because the wavelength is short, there are few sampling points, under the 44.1khz sampling rate, a 20 kHz wave only has three sampling points, when converting to a higher frequency, more points will be inserted, and the original appearance should be kept as much as possible. How to insert this point is a very difficult algorithm. For example

Assume that the triangle is a wave of the length of one second, and the sampling rate is 3Hz. Now we need to SRC to 4Hz. The only thing we can do is the timeline (horizontal) divide 4 points, take the vertical line of the point and the triangle edge interchange value, this process is re-sampling, the result is changed to a trapezoid. In the waveform diagram, the vertical axis corresponds to the energy value of the wave, which means that the signal strength of the wave is weakened and attenuation occurs. This example shows that the frequency conversion of non-integer times will change the waveform, which is inevitable. If the algorithm is good, it is possible to ensure that the converted waveform is similar to that before conversion, however, there are very few good algorithms, and most of the existing sound card SRC algorithms are very bad. As in the above example, high-frequency attenuation is caused by Src, and SRC may cause some other problems, for example, increasing inter-adjustment Distortion

Time Domain sampling theorem

Band:FContinuous signalF(TA series of discrete sample values are available.F(T1 ),F(T1 ± DeltaT),F(T1 ± 2 △T),..., As long as the interval between these sampling points isT≤ 1/2FThe original signal can be completely restored according to the sample values.F(T)

 

What is Crystal?

Crystal Chinese name is a Z crystal oscillator or crystal oscillator. Common sound cards support signal input of 44.1khz and 48 kHz, and the two sampling rates are not integer multiples. If both sampling rates are supported, two crystal crystals are required on the sound card. The values are 12.288mhz and 24.576 MHz (or an integer multiple of them ).

Audio sampling ad

Digital audio systems reproduce the original sound by converting the acoustic waveform into a series of binary data. The device used in this step is A/D) it Samples sound waves at tens of thousands of times per second. Each sampling record the state of the original simulated sound waves at a certain time point, which is called a sample. Connect a series of samples to describe a piece of acoustic waves. The number of samples per second is called the sampling frequency or sampling rate, in Hz ). The higher the sampling frequency, the higher the acoustic frequency that can be described. The sampling rate determines the sound frequency range (equivalent to the tone), which can be expressed in digital waveforms. Frequency expressed in Waveforms
The rate range is generally referred to as bandwidth. Correct understanding of audio sampling can be divided into the number of sampling digits and the sampling frequency.

Sampling precision

The number of sampling digits can be understood as the resolution of the sound processed by the acquisition card. The larger the value, the higher the resolution, and the more real the recording and playback sounds. First, we need to know that the sound files in the computer are represented by numbers 0 and 1. Therefore, the essence of recording on a computer is to convert analog sound signals into digital signals. On the contrary, the digital signal is restored to analog sound signal output during playback. The position of the acquisition card refers to
The binary digits of the digital sound signal used to play the audio file. The position of the acquisition card objectively reflects the accuracy of the digital sound signal on the input sound signal description. 8 represents the Power 8 of 2-, and 16 represents the power 16 of 2-64 K. Compared with a piece of same music information, a 16-bit sound card can be divided into 64 K precision units for processing, while an 8-bit sound card can only process 256 precision units, this results in greater signal losses, and the final sampling effect is naturally incomparable.

Audio sampling frequency fs

Digital audio systems reproduce the original sound by converting the acoustic waveform into a series of binary data. The device used in this step is A/D) it Samples sound waves at tens of thousands of times per second. Each sampling record the state of the original simulated sound waves at a certain time point, which is called a sample. Connect a series of samples to describe a piece of acoustic waves. The number of samples per second is called sampling.
Frequency or collection rate, in Hz ). The higher the sampling frequency, the higher the acoustic frequency that can be described. Sampling frequency refers to the number of times the recording device samples the sound signal within one second. The higher the sampling frequency, the more real the sound is restored. In today's mainstream acquisition cards, the sampling frequency is generally divided into three levels: 22.05 kHz, 44.1 kHz, and 48 khz. 22.05 kHz can only achieve the sound quality of fmbroadcast, 44.1khz is the theoretical limit of CD sound quality, while 48 khz is more accurate. Human ears cannot identify the sampling frequency higher than 48 khz
So there is little use value on the computer.

WAV audio format file size and playback time Conversion

The sampling rate is generally 44.1 kb with 16-bit sampling accuracy. The data is stored in WAV format. The value is 44.1 kHz (sampling rate) x 16-bit (sampling precision) x 2 (dual-channel) x playback time.

The WAV format is not compressed and lossless. The MP3 format is compressed and saved at, so the MP3 format size is equal to 1/12 of the above format.