Sound has always been the food of human spirit, and the voice module is a recording chip module with playback function and storage function; If you want to play complex and diverse sounds without distortion, a good voice format is very important. So what are the common voice module formats? The small braid of nine core electronics will take you to have a look.

▲NR7100s

PCM file format:

Pulse code modulation single pulse number demodulation, which samples the sound analog signal to obtain the video voice statistical data after quantitative analysis. It is the basic video voice file format. It is also very similar to raw file format and snd file format. They are all pure voice format.

Wav file format:

Wave audio files is a sound format file developed and designed by Microsoft, also known as wave sound files. It is widely supported by windows service platform and mobile phone applications. WAV format supports many compression algorithms and supports a variety of audio bits, sampling frequencies and channels. However, the wav format requires too much storage space to facilitate communication and transmission. Each data stored in the wav file has its own identification information. According to these identification information, the frequency and number of samples, single or stereo, etc. can be communicated to the user.

ADPCM format:

The past sample values are used to predict the current input sample value, the adaptive prediction function is compared with the actual detection value, and the measured difference is automatically quantified to make it always change synchronously with the signal. It is suitable for the case of moderate sound change rate and short sound playback process. Its advantage is the realistic processing of human voice, which is generally more than 90%, and 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. When encoding, first analyze the spectrum of the audio file, then filter out the noise level through the filter, and then discretely quantize and arrange each remaining bit to form an MP3 file with higher compression ratio. The compressed file can achieve the sound effect closer to the original sound source during playback. Its essence is that VBR (variable bit rate) can dynamically select the appropriate baud rate according to the content of the number, so the result of the number is to ensure the timbre and take care of the size of the document. The compression ratio of MP 3 is 10x to 12x. The first high-pressure compression rate sound format.

Linear scale file format:

According to the change rate of sound, the sound is divided into several segments, which are reduced by linear proportion, but its proportion is changeable.

Logpcm file format:

Most carry out linear reduction on all sounds and remove the final bits. This kind of reduction method is very easy to establish in hardware configuration, but the timbre is somewhat worse than linear scale. The actual effect is weak when the sound is small and the sound is detailed. It is mainly used for pure speech.