PCM format:

Pulse code modulation is pulse code modulation. It samples the sound analog signal to obtain the quantized speech data. It is a basic original speech format. Very similar to it are raw format and snd format. They are all pure voice format.

WAV format:

Wave audio files is a sound file format developed by Microsoft, also known as wave sound files. It is widely supported by Windows platform and its applications. WAV format supports many compression algorithms and supports a variety of audio bits, sampling frequencies and channels. However, wav format requires too much storage space and is not convenient for communication and propagation. Each piece of data stored in the wav file has its own independent identification, which can tell the user what data it is. These data include sampling frequency and bits, mono or stereo, etc.

ADPCM format:

It uses the past several sampling values to predict the current input sample value, makes it have adaptive prediction function, compares it with the actual detection value, and automatically quantifies the difference at any time, so as to keep it changing synchronously with the signal. It is suitable for the situation of moderate voice change rate, and the sound playback process is short. Its advantage is that the processing of human voice is more realistic, generally more than 90%. It has been widely used in the field of telephone communication.

MP3 format:

Moving Picture Experts Group Audio Layer III, referred to as MP3 for short. It uses the technology of MPEG Audio Layer 3 and adopts a coding algorithm called "sensory coding technology": during coding, first analyze the spectrum of the audio file, then filter out the noise level with a filter, and then scatter and arrange the remaining bits by quantization to form an MP3 file with high compression ratio, And the compressed file can achieve the sound effect closer to the original sound source when playing back. Its essence is that VBR (variable bit rate) can dynamically select the appropriate baud rate according to the coding content. Therefore, the coding result not only ensures the sound quality, but also takes care of the file size.

MP3 compression ratio is 10 times or even 12 times. It is a new voice format with high compression rate.

Linear scale format: according to the change rate of sound, the sound is divided into several segments, and each segment is compressed with linear proportion, but its proportion is variable.

Logpcm format:

Basically, the whole sound is linearly compressed and the last few bits are removed. This compression method is easy to implement in hardware, but the sound quality is worse than linear scale, especially when the volume is small and the sound is delicate. It is mainly used for pure speech.