Audio sampling-Explanation

Digital audio systems reproduce the original sound by converting the acoustic waveform into a series of binary data. The device used in this step is A/D) it Samples sound waves at tens of thousands of times per second. Each sampling record the state of the original analog sound waves at a certain time point, which is called a sample. Connect a series of samples to describe a piece of acoustic waves. The number of samples per second is called the sampling frequency or sampling rate, in Hz ). The higher the sampling frequency, the higher the acoustic frequency that can be described. The sampling rate determines the sound frequency range (equivalent to the tone), which can be expressed in digital waveforms. The frequency range expressed in waveforms is generally referred to as bandwidth. Correct understanding of audio sampling can be divided into the number of sampling digits and the sampling frequency.

1. Number of sampling digits

The number of sampling digits can be understood as the resolution of the sound processed by the acquisition card. The larger the value, the higher the resolution, and the more real the recording and playback sounds. First, we need to know that the sound files in the computer are represented by numbers 0 and 1. Therefore, the essence of recording on a computer is to convert analog sound signals into digital signals. On the contrary, the digital signal is restored to analog sound signal output during playback. The bit of the acquisition card is the binary number of the digital sound signal used by the acquisition card to collect and play audio files. The position of the acquisition card objectively reflects the accuracy of the digital sound signal on the input sound signal description. 8 represents the Power 8 of 2-, and 16 represents the power 16 of 2-64 K. Compared with a piece of same music information, a 16-bit sound card can be divided into 64 K precision units for processing, while an 8-bit sound card can only process 256 precision units, this results in greater signal losses, and the final sampling effect is naturally incomparable.
Today, all mainstream products on the market are 16-bit acquisition cards, not the 64-bit or even 128-bit cards that some ignorant merchants advocate, they confused the concept of the secondary audio of the acquisition card with the concept of the number of sampling digits. Although the emu10k1 chip, which is currently the most powerful Collection Card series, claims to be 32-bit, it is only a multi-audio stream technology based on direct sound acceleration, in essence, it is a 16-bit sound card. It should be said that the 16-bit sampling accuracy is more than enough for computer multimedia audio.

2. Audio sampling level (audio sampling frequency)

Digital audio systems reproduce the original sound by converting the acoustic waveform into a series of binary data. The device used in this step is A/D) it Samples sound waves at tens of thousands of times per second. Each sampling record the state of the original analog sound waves at a certain time point, which is called a sample. Connect a series of samples to describe a piece of acoustic waves. The number of samples per second is called the sampling frequency or sampling rate, in Hz ). The higher the sampling frequency, the higher the acoustic frequency that can be described.
Sampling frequency refers to the number of times the recording device samples the sound signal within one second. The higher the sampling frequency, the more real the sound is restored. In today's mainstream acquisition cards, the sampling frequency is generally divided into three levels: 22.05 kHz, 44.1 kHz, and 48 khz. 22.05 kHz can only achieve the sound quality of fmbroadcast, 44.1khz is the theoretical limit of CD sound quality, while 48 khz is more accurate. Human ears can no longer identify the sampling frequency higher than 48 khz, so there is little value for use on the computer.

3. Bit Rate description
Bit Rate refers to the amount of information that can be passed per second in a data stream. You may have seen the situation described in the audio file "128-kbps MP3" or "64-kbps WMA. Kbps indicates the number of kilobytes per second. Therefore, a larger value indicates more data:-kbps MP3 audio files contain twice the data volume of-kbps WMA files and occupy twice the space. (But in this case, the two files sound no different. Why? Some file formats can use data more effectively than other files. The sound quality of the 64-kbps WMA file is the same as that of the 128-kbps MP3 file .) It is important to know that the higher the bit rate, the larger the amount of information, the larger the processing capacity for decoding the information, and the more space the file occupies.
Selecting an appropriate bit rate for the project depends on the playback target: If you want to put the VCD to the DVD player for playback, the video must be 1150 kbps, and the audio must be 224 kbps. A typical 206 MHz Pocket PC-supported MPEG Video can reach 400 kbps-an exception occurs when the playback limit is exceeded.

Psychological acoustics audio compression
The word "psychological acoustics" seems confusing. In fact, it is very simple. It refers to "the way the human brain interprets sounds ". All formats of compressed audio use powerful algorithms to remove audio information that we cannot hear. For example, if I scream and tap your feet, you will hear me, but you may not hear me. By removing the stepping sound, the information will be reduced and the file size will be reduced, but it sounds no different.