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The principle of sound
Sound is a kind of sound wave produced by vibration, which is transmitted through medium (air or solid or liquid) and can be perceived by human or animal auditory organs. The frequency of sound is generally expressed in Hertz, and is recorded as Hz, which refers to the number of periodic vibrations per second. Decibels are units used to represent sound intensity, which is recorded as dB.
Sound is a kind of fluctuation. When playing an instrument, beating a door or knocking on the table, the vibration of sound will cause the rhythmic vibration of the medium air molecules, which causes the air around to change in density and form a dense and dense longitudinal wave, which produces sound waves, which will continue until the vibration disappears.
The frequency of sound received by any organ has its range limitation. Human ears generally only hear sounds in the range of 20Hz to 20000 Hz (20kHz), and the upper limit will decrease with the increase of age. Other species also have different auditory frequencies, such as dogs that can hear sounds over 20kHz but not below 40Hz. The range of hearing frequencies of different species of animals is as follows:
① Bat: 1000-120000hz
② Dolphin: 2000-1000000hz
③ Cat: 60-65000hz
④ Dog: 40-50000hz
⑤ Person: 20-20000hz
⑥ Red: infrasound, blue: audible sound, green: Ultrasonic
1. Microphone acquisition
Microphone (also known as microphone or microphone, officially called microphone in Chinese), translated from English microphone, is a transducer that converts sound into electronic signal. According to the principle of microphone making, it can be divided into the following categories:
(1) Moving microphone
The basic structure of the dynamic microphone consists of coil, diaphragm and permanent magnet. When sound waves enter the microphone, the diaphragm vibrates under the pressure of sound waves. The coil connected with the diaphragm starts to move in the magnetic field. According to Faraday law and Lenz law, the coil will generate induction current.
Because of the coil and magnet, the dynamic microphone is not light and sensitive, and the high and low frequency response is poor. The advantage is that the sound is more soft and suitable for recording human voice.
1. Sound wave 2. Vibration film 3. Coil 4. Magnet 5. Output signal
(2) Condenser microphone
There is no coil or magnet in the condenser microphone, and the voltage change is generated by the change of the distance between the two plates of the capacitor. When the sound wave enters the microphone, the vibration film vibrates, because the substrate is fixed, so that the distance between the vibration film and the substrate will change with the vibration. According to the characteristics of the capacitance, when the distance between the two partitions changes, the capacitance value C will change, and the power Q will be changed when the C changes. Because the fixed plate voltage V is needed in the condenser microphone, additional power is needed for this microphone to operate. The common power supply is battery. Because of its high sensitivity, capacitance microphone is often used for high-quality recording.
1. Acoustic wave 2. Vibration film 3. Substrate 4. Battery 5. Resistance 6. Output signal
(3) electret condenser microphone
The condenser microphone usually needs additional power supply to operate, but the electret condenser microphone can not need additional power. Electret is also called "permanent electric body", which will have a fixed number of charges. The whole line has no power consumption (the line removes the battery and resistance shown in the figure above). According to the formula: q = Cu, when C changes, the voltage u at both ends of the capacitor will inevitably change, thus outputting the electrical signal to realize the sound electricity transformation. Because the actual capacitor has a small capacitance, the output electrical signal is very weak, the output impedance is very high, which can reach more than 100 megaohms. Therefore, it can not be directly connected with the amplifier circuit, and it must be connected with impedance converter. A special field effect tube and a diode are usually used to form impedance converters. Because field effect tube is an active device, it needs a certain bias and current to work in the amplification state. Therefore, a DC bias is required to be added to the electret microphone to work.
(4) MEMS microphone
MEMS microphone refers to a microphone made of MEMS technology, also known as microphone chip or silicon microphone. The pressure sensing film of MEMS microphone is etched on silicon chip directly by MEMS technology. The IC chip is usually integrated into some related circuits, such as preamplifier. Most of the MEMS microphone design is a kind of change of capacitor microphone in basic principle. The MEMS microphone also often has an analog-to-digital converter, which can directly output digital signals and become a digital microphone, so as to connect with the current digital circuit. The MEMS microphone is mainly used in some small mobile products such as mobile phones and PDAs.
There are other types of microphones that don't talk much about here.
2. Microphone noise reduction
With the development of technology, even in a very noisy environment, the other side can hear the phone clearly, which is mainly due to the development of noise reduction technology. In the current mobile phones, we often see that there are not only one microphone, but two or even three, and the key to noise reduction is the more.
(1) Microphone noise reduction
Generally speaking, the phone has two microphones, one at the top and one at the bottom. Both look very small, but the two have a distinct difference, where the bottom is used to provide clear calls, while the top is used to eliminate noise.
Because the distance between the top and bottom is different from the source of the voice during the call, the volume volume picked up by the two wheat is different. With this difference, we can filter out the noise and keep the human voice. When making a call, the background noise volume picked up by the two microphones is basically the same, while the recorded voice will have a volume difference of about 6dB. After the top wheat collects noise, it can be used to eliminate noise after generating compensation signal by decoding.
(2) Echoes
Echo (or echo) refers to the reflection of the sound by obstacles. When an obstacle is encountered, one part of the sound waves pass through the obstacle, while the other will reflect back to form an echo. If the obstacle has a hard and smooth surface, it is easy to generate echo; otherwise, it is easy to absorb sound with soft surface; in addition, rough surface is easy to scatter sound. Echo is longer than those that are directly transmitted, so it is heard later than the direct sound. If the interval between two lines of sound waves is less than 0.1 seconds, the human ear can not distinguish, and only the extended sound can be heard. Because the sound velocity in the gas is 343 meters per second at room temperature (20 ℃), people standing at the sound source need to hear the echo, and the distance from the obstacle to the sound source is at least 17 meters.
(3) Echo cancellation
Many times, there is a demand for connecting wheat to live broadcast, and the echo cancellation of the collected sound is needed. When the mobile phone is in the situation of connecting wheat, the mobile phone plays the other party's voice, collects it with microphone, and then transmits the collected sound to the other party. In this way, the other party will hear its own echo. Because the loop is going on all the time, the echo will be more and more, and finally there will be a buzz.
Echo cancellation is to remove the voice played by the phone itself when recording the external sound of the microphone, so that the other party's voice is filtered out of the collected sound, thus avoiding the echo generation. The following picture shows the mechanism of echo cancellation.
Echo cancellation
At the near end, the microphone will collect the remote sound from the speaker. Suppose the sound is y (n). Of course, because it is necessary to broadcast the remote sound, we can certainly get the sound signal from the remote end, assuming that the sound is x (n). It is not difficult to find that x (n) is played by speakers, then transmitted by air, and finally collected by microphone, and then changed to y (n), X (n) and Y (n) have obvious correlation. Assuming that the total sound signal collected by microphone is Z (n), y (n) in Z (n) needs to be found by adaptive filter according to X (n), and then y (n) is filtered out from Z (n).
3、 Sound acquisition
The principle of microphone has been described before. After the microphone is collected into sound, it is converted into analog electrical signal. After that, it is necessary to convert analog electrical signal into analog signal recognized by computer.
Audio record can be used in Android to record sound, and recorded sound can be set as PCM sound. To express sound in computer language, it is necessary to digitize sound. The most common way to digitize sound is to modulate PCM (pulse code modulation) by pulse code. Sound passes through the microphone and converts it into a series of voltage changing signals. To convert such voltage changing signal into PCM signal, three processes are needed: sampling, quantification and coding. To implement these three processes, three parameters are needed: sampling frequency, number of sampling bits and number of channels.
Pulse Code Modulation
(1) Sampling frequency
Sampling frequency is the sampling frequency, which refers to the number of times that sound samples are obtained every second. The higher the sampling frequency, the better the quality of sound, the more real the sound restoration is, but it also takes up more resources. Because the resolution of human ear is very limited, too high frequency can not be distinguished. There are 22khz, 44KHz and other levels in 16 bit sound cards, among which 22khz is equivalent to the sound quality of ordinary FM broadcasting, 44KHz is equivalent to CD sound quality, and the current commonly used sampling frequency is no more than 48Khz.
(2) Sample number
The number of sampling bits is the sampling value or the sampling value (that is, the sample amplitude is quantified). It is a parameter used to measure the fluctuation of sound, or the resolution of sound card. The larger the value, the higher the resolution, the stronger the ability of the sound produced.
In computer, the sampling number is generally divided into 8 bits and 16 bits. 8 bits are not to say that the vertical coordinates are divided into 8 parts, but are divided into 8 times of 2, namely 256; the same reason 16 bits divide the vertical coordinates into 65536 parts of the 16 order of 2.
The larger the sampling rate and the sample size, the more the recorded waveform is closer to the original signal.
(3) Number of channels
It is very well understood that there is a division of mono and stereo, and mono sound can only be made by one speaker (some of which can also be processed as two speakers output the same sound channel). The PCM of stereo can make both speakers sound (generally, there is division of labor between left and right channels), and it can feel more spatial effect.
So, now we can get the formula of the capacity of PCM file:
Storage quantity = (sampling frequency, sampling number, channel time) / 8 (unit: bytes)
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