What do the bits, bit rate and sample rate of an audio file mean?
For example, the common mp3 format audio source
In order to store a continuous physical signal (well, tell me about Planck’s constant…) in a computer, it must be converted to a digital signal. In acoustics, a digital signal is a digital representation of the amplitude of the sound wave at any moment.
Sound waves are longitudinal waves, which are difficult to draw. The following figure is replaced by transverse waves (the concept of longitudinal waves is the phenomenon that the density of air or other media changes regularly due to energy. The peaks represent high density, the troughs represent low density, and the horizontal line is the average density, i.e. silent state)
Using high school physics, waves contain two dimensions, one is intensity and the other is time. “Number of digits” indicates how many levels sound waves are divided into from the strongest to the weakest; “Sampling Rate” determines the precision of the time axis or the sampling density, that is, the length of time represented by each red dot, and the code rate is one second The number of dots on the clock, multiplied by the space that each point occupies.
So the so-called 24 bits consist of dividing the intensity of the sound wave by 2 at power level 24, occupying 3 bytes of space. Obviously, the finer the grade, the more details are restored.
The sample rate is generally 44100 Hz for CD (Hertz = times/second), 48000 Hz for DVD, and 96000 Hz as standard. As with the number of digits, the more points you get in a single second, the more details you retrieve. Why does CD take this value? Because the hearing range of the human ear is generally believed to be between 20 and 20,000 Hz. A peak and a trough need to be represented, and at least two sampling points are required. Therefore, the CD can represent the sound of 22050 Hz at most, but this sound does not have any detail, because if there are only two peak and valley points, the average waveform is completely lost. Therefore, there will be a higher sampling rate.
If it’s in a lossless uncompressed format, the bit rate is strictly equal to the number of bits * sample rate * number of channels. And typically, the MP3 bitrate you can see just represents how much capacity the format needs to describe this one second of audio.
MP3 is lossy compression. In the compression process, some information is lost, but the lost information cannot be represented by the number of bits and the sampling rate. Generally, the higher the code rate, the less information is lost. Mathematically, bitrate and sound quality are proportional. As for whether you can hear it or not, it depends on many factors. The MP3 algorithm is not complicated, of course, to understand it you have to learn what the Fourier transform is.
There is also lossless compression (representing APE, FLAC, etc.), which also has a bitrate, and this bitrate has nothing to do with sound quality. It also describes how much capacity the file uses to describe one second of audio content, but the same audio content can be compressed to different sizes (compression ratios), similar to zip compression ratios. No matter how big you compress it, in the end it can be restored to the same file. So if you see someone looking for a lossless bitrate, you can basically conclude that the product is a bad pen.
