What is the difference between 128k and 320k music? Part 2
In simple terms, we can think of a sound wave as a curve.
We know that the curve is made up of points, and the sampling rate is the number of points in the middle of the length per second (the horizontal axis in the figure above). Sampling precision is the number of points in the dynamic range (upper vertical axis). The finer the positioning of these two dimensions, the greater the true sound restoration and the better the sound quality. Of course, the larger the audio file will be. The customer mentioned by the above colleague said that the latest Hi-Res Audio format released by SONY is a 6-channel 192kHz/24-bit recorded audio file. The size of the lossless format, of course, will be more than 200 megabytes.
The sampling frequency is approximately the following depending on the type of use (k is the thousand-bit symbol, 1khz=1000hz):
8khz: used for phones etc, is enough to record human voices.
22.05khz: transmission use frequency.
44.1kb: Audio CD.
48khz: used in DVD and digital TV.
96khz-192khz: used for DVD-Audio, Blu-ray HD, etc.
The common range of sample precision is 8 bits to 32 bits, with 16 bits generally used on CD.
Having said that, my friends are starting to get confused. It’s not the bitrate that determines the sound quality, so why is everyone saying that 320kb sound quality is better than 128kb?
【Audio Compression】
Well, in fact, the bit rate should be said to be another dimension, it is a compression of audio files.
Nowadays, most of the audio formats we use regularly are based on the original “WAV” file of the audio CD (44.1khz sample rate, 16bit sample precision, 2ch). The original recorded sound data is stored in a matrix, which is in PCM format, while WAV format is an encoding format developed by Microsoft. Its function is to reproduce the data in PCM format through encoding.
Since the data in WAV basically completely restores the PCM data, MP3, AAC and other lossless encoding formats are basically recompressed based on the WAV files. Therefore, we can simply think that WAV is the original audio format and other audio formats are compressed formats.
When it comes to compression, storage and transmission are inseparable. The purpose of compression is to improve storage and transmission, so before we talk about compression, we need to understand the basic units of computers.
We all know that the computer is a binary number system, and the files stored by the computer are made up of two numbers, 0 and 1. Therefore, the computer’s transmission is based on each number, and each number is called 1 ” bit”. For example, for an audio piece, its basic data is “0,1,1,1,0,1, 1 ,0”, and when transmitting, these numbers are transmitted one by one. The sampling precision mentioned above is this unit.
The storage unit of the computer is “byte (Byte)”. In the computer, 1 byte consists of 8 bits, that is, 8b(bit)=1B(Byte). In computer parlance, data storage is expressed in decimal and data transmission is expressed in binary, so 1KB=1024B=1024×8b. This is also part of the reason why the hard drive capacity we see does not match the actual capacity.
Go back and talk about audio compression, the bitrate of the audio is actually the compression ratio. So the bitrate really just defines the size of the file, but because under normal conditions the larger the file, the less data you lose, so the sound quality is relatively higher. However, the bit rate itself does not directly affect the quality of the file. For example, if we take a 128kb file as the source file, even if it is converted to a 320kb file, the sound quality will not be better than 128kb. .
