Analysis of sound coding technology for multimedia technology applications

Foreword

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There are several main aspects of multimedia data types:

Graphics and images: images consist of pixels; graphics consist of primitives

Text: The most basic, there are many encoding methods, ASCII code, Chinese GB code, etc.

Audio: The WAV file is the most basic form of sound recording in a computer. It samples the sound wave signal and quantizes it to form a digital signal. The actual data recording in MIDI format is "score" (digital audio and video technology).

Animation and video: Animation is the use of a computer to generate a series of continuous screen technologies for real-time presentation. The video consists of a sequence of real pictures taken from a frame.

Usually, we will use computer technology to integrate text, images, graphics, animation, audio and video into a variety of media, so that they can establish a logical connection, and can acquire, encode, edit, and process them. The technology of storage, transmission and reproduction is called multimedia technology. In practical life, the application of multimedia technology is very extensive. The following is a small example of audio coding technology in multimedia technology (audio and video technology), which introduces the relevant knowledge about multimedia technology applications in real life.

1.G.721 ADPCM encoder

ADPCM is a waveform coding technique that uses a high correlation between samples and samples and quantization step adaptation to compress data.

CCITT has developed the G.721 recommendation for this standard, which is called 32kb/s Adaptive Pulse Code Modulation to 24and 40kb/s for Digital Circuit Multiplication Equipment Application. The data rate of the encoder using this standard can be reduced to 40kb. /s and 24kb/s.

The G.721 ADPCM standard recommended by CCITT is a transcoding system. It uses ADPCM conversion technology to convert between 64kb/s A-law or μ-law PCM rate and 32kb/s rate.

2. Subband coding (SBC)

The main process of subband coding is:

1 A set of band-pass filters (BPF) is used to divide the frequency band of the input audio signal into several consecutive frequency bands, each of which is called a sub-band.

2 The audio signal in each subband is coded using a separate coding scheme.

3 When transmitting on the channel, the code of each sub-band is combined.

4 When decoding at the receiving end, the code of each sub-band is decoded separately, and then they are combined and restored to the original audio signal.

The benefits of separately encoding each subband:

First, each sub-band signal is adaptively controlled, and the quantization step can be adjusted according to the energy level of each sub-band. Subbands with higher energy levels are quantized with a large quantization step to reduce the total quantization noise.

• Second, each sub-band can be assigned a different number of bits according to the perceptual importance of each sub-band signal to represent each sample value. For example, in the low frequency sub-band, in order to protect the structure of the pitch and the formant, a smaller quantization step and a larger number of quantization steps are required, that is, a larger number of bits are allocated to represent the sample value. Frictional and noise-like sounds in speech usually appear in high frequency subbands, allocating fewer bits to it.

3. Subband-Adaptive Differential Pulse Code Modulation (SB-ADPCM)

The G.711 standard with a sampling rate of 8 kHz, 8 bits/sample, and a data rate of 64 kb/s is a compiled code standard developed by CCITT for a voice signal frequency of (300 to 3400) Hz, which is a narrowband audio signal coding. Modern voice coding technology has been able to reduce data rates without significantly degrading sound quality. CCITT's recommended 8KHz sampling rate, 4 bits/sample, 32 kb/s G.721 standard, and G.721 extended standard G.723 all illustrate the progress of voice compression coding technology.

G.722 is the audio signal coding and decoding standard recommended by CCITT. The standard is a codec principle, algorithm, and computational details that describe an audio signal bandwidth of 7 kHz and a data rate of 64 kb/s.

The main goal of G.722 is to maintain a data rate of 64 kb/s, and the quality of the audio signal is significantly higher than the quality of G.711. The G.722 standard increases the audio signal sampling frequency from 8 kHz to 16 kHz, which is twice the sampling rate of G.711 PCM, so the frequency of the signal to be encoded is extended from the original 3.4 kHz to 7 kHz. This greatly improves the quality of the audio signal, from the voice quality of digital telephones to the quality of amplitude modulated (AM) radio broadcasts. In terms of the quality of the speech signal, there is not much improvement in the sampling rate, but the quality of the signal such as music is greatly improved.

The G.722 codec system uses its own adaptive differential pulse code modulation technique. The sub-generations that divide the frequency band into two equal bandwidths are the high frequency sub-band and the low frequency sub-band. The signals in each of the equal bandwidth subbands are encoded with ADPCM.

4.G.722 SB-ADPCM codec

In order to meet the growing needs of video teleconferencing, CITT developed the G.722 recommendation for this purpose in 1988, called "7KHz audio signal coding with a data rate of 64kb/s - 7kHz Audio-coding with 64kb/s. The standard improves the quality of voice signals from telephone quality to AM radio broadcast quality, while its data transmission rate remains at 64 kb/s.

Broadband voice refers to voices with a bandwidth of (50-7000) Hz. This kind of voice has significantly improved intelligibility and naturalness compared to the bandwidth (300~3400) Hz, and it is easier to recognize the other party's speech. people.

5. Linear Predictive Coding (LPC)

Linear predictive coding is a very important coding method. In principle, LPC is a parameter that generates vocal stimuli and transfer functions by analyzing voice waveforms. The encoding of sound waveforms is actually converted into encoding of these parameters, which greatly reduces the amount of sound data. The parameters obtained by the LPC analysis are used at the receiving end to reconstruct the speech by the speech synthesizer.

The synthesizer is actually a discrete time-varying time-varying linear filter that represents the human voice generation system model. Time-varying linear filters are used both as predictors and as synthesizers. When analyzing voice waveforms, it is mainly used as a predictor. As the voice waveform changes, the model parameters and excitation conditions are periodically adapted to the new requirements.

The application prospects and prospects of multimedia technology

Prospects for multimedia technology applications

Family education and personal entertainment are the mainstream of the current international multimedia market

Content presentation and management information systems are important aspects of multimedia technology applications

·Multimedia communication and distributed multimedia systems are the future development direction of multimedia technology

The development direction of multimedia technology application:

The first is the multimediaization of the computer system itself;

Second, multimedia technology and on-demand TV, intelligent home appliances, identification network communication and other technologies are combined to make multimedia technology enter education, consulting and entertainment. Areas such as business management and office automation;

Third, multimedia technology and control technology penetrate each other and enter the field of industrial automation measurement and control.

to sum up

The real multimedia technology involves the object of computer technology, while other simple things, such as movies, television, audio, etc., are not in the category of multimedia technology. This paper takes sound coding technology as an example to introduce in detail the relevant knowledge about multimedia technology application in real life and a simple analysis of the future development of multimedia technology application.

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