Will audio compression technology increase the stream bit rate?

Audio compression technology will increase the stream bit rate; audio compression technology is to apply appropriate digital signal processing technology to the original digital audio signal stream. Audio compression is necessary to reduce the transmission bit rate of the original sound data. Audio compression, it will increase the audio stream bit rate after compression to improve the sound quality level.

The operating environment of this tutorial: Windows 10 system, DELL G3 computer.

Will audio compression technology increase the stream bit rate?

Yes.

Audio compression technology refers to the application of appropriate digital signal processing technology to the original digital audio signal stream (PCM encoding) without losing the amount of useful information or causing negligible losses. Reduce (compress) its code rate, also known as compression encoding. It must have a corresponding inverse transformation, called decompression or decoding. After the audio signal passes through a codec system, a large amount of noise and certain distortion may be introduced. Audio compression technology is to apply appropriate digital signal processing technology to the original digital audio signal stream. Audio compression is to reduce the original sound. Audio compression is only required based on the data transmission bit rate. After compression, it will increase the audio stream bit rate to improve the sound quality level.

Extended knowledge

Compression coding method

According to different compression principles, audio signal coding is divided into waveform coding, parameter coding and coding forms that integrate multiple technologies.

(1) Waveform coding directly samples the time domain or frequency domain waveform of the audio signal at a certain rate, and then quantizes the amplitude samples in layers and transforms them into digital codes. The waveform data produces a reconstructed signal encoding system derived from the original sample of the signal. The waveform is as consistent as possible with the original sound waveform, retaining the detailed changes and various transition characteristics of the signal.

(2) Parameter coding first establishes feature models based on different signal sources, such as language signals, natural sounds, etc., and strives to keep the reconstructed sound signal as high as possible through extraction of feature parameters and coding processing. The semantic meaning of the sound, but the waveform of the reconstructed signal may be quite different from the waveform of the original sound signal. Commonly used characteristic parameters include formants, linear prediction coefficients, frequency band division filters and other parametric encoding technologies that can achieve low-rate sound signal encoding. The bit rate can be compressed to 2Kbit/s - 4.8Kbit/s, but the sound quality can only reach Medium, especially with low naturalness, is only suitable for the transmission and expression of language.

(3) Hybrid coding The coding form that combines waveform coding and parametric coding overcomes the weaknesses of the original waveform coding and parametric coding, and strives to maintain the high quality of waveform coding and the low rate of parametric coding. In 4 - High-quality synthesized sound signals can be obtained at a rate of 16Kbit/s. The basis of hybrid coding is Linear Predictive Coding (LPC), and commonly used coding methods such as Pulse Excited Linear Predictive Coding (MPLPC), Planning Pulse Excited Linear Predictive Coding (KPELPC), Codebook Excited Linear Predictive Coding (CELPC) and other coding methods.

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