Without compression, the calculation formula of sound data volume is:

Data volume (bytes/second) = (sampling frequency (Hz)× sampling bit)× channel number)/8

The sound processing quality of a sound card can be measured by three basic parameters, namely, sampling frequency, sampling bit number and channel number.

2 sampling frequency refers to the number of sampling times per unit time. The greater the sampling frequency, the smaller the interval between sampling points, and the more realistic the digitized sound, but the larger the corresponding data volume. Sound cards generally provide different sampling frequencies such as 11.25kHz, 22.5kHz and 44.1kHz.

3 number of sampling bits is the number of bits to record the numerical value of each sampling value. The number of sampling bits is usually 8bits or 16bits. The larger the number of sampling bits, the finer the variation of sound that can be recorded, and the larger the corresponding data volume.

the number of four channels refers to whether the processed sound is mono or stereo. Monochannel has only one data stream in sound processing, while stereo needs two data streams of left and right channels. Obviously, the stereo effect is better, but the corresponding data volume is twice that of mono.

extended materials

AAC is actually the abbreviation of advanced audio coding. AAC is composed of Fraunhofer IIS-A, Dolby and AT&; An audio format developed by T***, which is a part of MPEG-2 specification. The algorithm adopted by AAC is different from that of MP3, and AAC improves the coding efficiency by combining other functions.

AAC's audio compression algorithm far exceeds some previous compression algorithms (such as MP3, etc.). It also supports up to 48 audio tracks, 15 low-frequency audio tracks, more sampling rates and bit rates, multi-language compatibility and higher decoding efficiency. In a word, AAC can provide better sound quality on the premise that it is 3% smaller than MP3 files.

Digital audio has become the mainstream because of its excellent sound quality, lossless transmission and various editing and conversion, and it has been applied in all aspects.

The common MP3, WMA and OGG are called lossy compression. As the name implies, lossy compression is to reduce the audio sampling frequency and bit rate, and the output audio file will be smaller than the original file.

another kind of audio compression is called lossless compression, which can reduce the volume of the audio file on the premise of 1% preservation of all the data of the original file. After the compressed audio file is restored, it can achieve the same size and the same bit rate as the source file.

lossless compression formats include APE, FLAC, WavPack, LPAC, WMALossless, AppleLossless, TTA, Tak, TAC, La, OptimFROG and Shorten, while common and mainstream lossless compression formats include APE, FLAC, TTA and TAK.

WAV CDs can capture music in this format. However, due to its large volume and uncompressed original audio, it can generally be compressed and converted into FLAC or APE with small volume. Note: wav still belongs to lossless format, while the latter two are lossless compression format.