MP3 is just an audio format.
And the audio has several important parameters, such as KHZ, BIT, channel, KBPS, etc.
And the format is different, the algorithm is also different, so even if the above parameters are the same, the format is different. The sound quality will also be very different. Among them, VBR is a dynamic sampling. For a detailed and complete explanation, please refer to the description below.
After reading it patiently, you will be able to say a thing or two.
Audio sampling explains the digital audio system The original sound is reproduced by converting the waveform of the sound wave into a series of binary data The equipment used in this step is an analog-to-digital (A/D) converter, which samples the sound wave at a rate of tens of thousands of times per second. A sampler records the state of the original analog sound wave at a given time, called a sampler. A series of samples can be connected to describe a sound wave. The number of samples per second is called the sample rate or sample rate, and the unit is HZ.
(hertz). The higher the sample rate, the higher the frequency of the sound wave that can be described. The sample rate determines the frequency range (equivalent to the pitch) of the sound, which can be represented by a digital waveform. The frequency range is often called the bandwidth. To understand properly, audio sampling can be broken down into the number of bits sampled and the frequency of the sample. 1. The number of sampling bits The number of sampling bits can be understood as the resolution of the sound processed by the capture card. The higher the value, the higher the resolution and the more realistic the sound recorded and played back. The first thing we need to know: sound files on the computer are represented by the numbers 0 and 1 . So the essence of recording on the computer is to convert the analog sound signal into a digital signal. On the contrary, during playback, the digital signal is restored to an analog sound signal output. The capture card bit refers to the binary digits of the digital sound signal used by the capture card when capturing and playing sound files. The bits on the capture card objectively reflect the accuracy of the digital sound signal’s description of the input sound signal. 8 bits represent the eighth power of 2–256 and 16 bits represent the sixteenth power of 2–64K . For comparison, for the same music data, a 16-bit sound card can break it down into 64K precision units for processing, while an 8-bit sound card can only handle 256 precision units.
It causes a large signal loss, and the final sampling effect is naturally incomparable. All the major products on the market today are 16-bit capture cards, not 64-bit or even 128-bit as some ignorant marketers advocate, who confuse the concept of polyphony in capture cards with the concept of bits of sampling. Although the EMU10K1 chip used by the most powerful capture card series is claimed to be 32-bit, it is just a Direct Sound acceleration-based multi-audio streaming technology, and its essence is still a 16-bit sound card. . It must be said that the 16-bit sampling precision is more than enough for computer multimedia audio. 2. Audio Sampling Level (Audio Sampling Rate) The digital audio system reproduces the original sound by converting the sound waveform into a series of binary data. The equipment used to accomplish this step is an analog-to-digital (A/D) converter, which takes each sound wave, and each sampler records the state of the original analog sound wave at a given time, which is called sample. A series of samples can be connected to describe a sound wave. The number of samples per second is called the sample rate or sample rate, and the unit is HZ (Hertz). The higher the sample rate, the higher the frequency of the sound wave that can be described.
