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It's Shannon's sampling theorem.
In order to ensure that the signal is not distorted, the sampling frequency should be "= 2 times the highest frequency of the signal.
That is: for example, if the highest frequency of the signal is f=1kHz (I don't remember how much the highest frequency of speech is, here is just an example), then the sampling frequency is at least 2kHz
If the signal frequency is up to 2kHz, then the sampling should be at least 4kHz
That should make sense.
If the sampling frequency is too low, the signal at high frequencies will be lost, which can also be said to be treble pitching, but it is not lost. It's not that there's no sound in the treble, but the treble is out of tune, understand?
Who's going to make that kind of software, that's boring!
However, there are other software that can be tried.
I remember there used to be an audio compression software that could compress in various formats**.
It is a CD sound quality, which can be compressed into WMA, RM ...... respectivelyVarious formats. The principle of his compression is to reduce the sampling frequency.
Because some **require the size of uploading**, that software is used to help**file "**". I've pressed it, and the difference in the effect of pressing it out is really big. The worst effect is the same as the sound quality of the mobile phone**, due to the low sampling frequency, you can only hear the other party's voice, you can tell who it is, but it is not pleasant.
The ** high sample of CD sound quality is very pleasant to the ear, which should be experienced.
I don't remember the specific name of that software, but the software I played a few years ago ......
Actually, there's no need to look for that software.
If you really want to listen to the changes in the downsampling frequency, there is a simple way: play it again with the stereo first, and then ask the home mobile phone to call you**, and listen to it again from the mobile phone**, you will know the effect.
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Yes, the higher the frequency, the shorter the period, if the sampling rate is not enough, this part will be skipped. Due to the long period of bass, the bass can be sampled from the peak to the trough to the median, and the bass is restored when it is restored, and the treble is lost. Just draw a picture and you'll find out.
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Answer] :d This question tests more than ** basic knowledge. Waveform sound information is a data sequence used to represent the amplitude of Fenghe sound, which is obtained by sampling the analog sound at a certain interval, and then quantized and encoded to obtain a data format that is convenient for computer storage and processing.
After the sound signal is digitized, its data transmission rate (bits per second) is directly related to the real-time transmission of the signal in the computer, and the total data volume is directly related to the computer storage space.
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According to Narquis's theorem, as long as the sampling frequency is twice the highest frequency of the original signal, the sound can be restored realistically. If you sample a 20k audio signal, 48k is enough, and the signal of digital audio recording is originally discrete, it is not like the analog signal is continuous, which is why you can record the original signal better at every point you see.
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Sorry to correct that the sample rate and the sound volume are not necessarily related. The sampling rate is simply the frequency in the digital sampling process of an analog signal. The highest frequency that a person can hear is 20kHz
Therefore, in order to record all the information that the human ear can hear, at least on a waveform of 20kHz, one vibration needs to have 2 sampling points. Let's put it this way, otherwise the waveform is sampled into a straight line, and there will be no information about this frequency. So the general sampling rate is.
Minimum sampling on a 20kHz waveform is guaranteed. Therefore, no matter how low the sampling rate is, there is no information recording on the high frequency. If it's purely recording vocals, it's okay, but other things are different. So try to keep the sampling rate high, such as 48kHz above
If you want it again, it doesn't make much sense to use it in ordinary times. Therefore, the sample rate has nothing to do with the volume, if you want to increase the volume, you can use the software to increase it after recording. A sample rate of 48kHz is recommended