1. Compression coding of digital audio signals
After digitizing the audio signal with linear PCM, the code rate is approximately 700 kbit/s. For example, the sampling rate of CD is 44.1 kHz, and each sample is represented by a 16-bit word length, and its code rate is 705.6 kbit/s, and the code rate of one stereo signal reaches 1411.2 kbit/s. Such a high code rate requires a wide frequency band for transmission and requires a large storage capacity for storage, which greatly limits the application of digital audio and makes digital audio broadcasting difficult. In order to solve this problem, it is necessary to adopt a compression coding technique to effectively reduce the code rate while basically ensuring close to the CD sound quality.
There are a variety of audio compression coding techniques applied in digital broadcasting, including digital television and digital audio broadcasting, which are based on the characteristics of the human ear to achieve a reduced rate. In these coding schemes, the frequency of the human ear, the temporal shadowing effect, and the localization characteristics of the sound are utilized. The frequency shadowing effect is that when two signals with close frequency and distinct intensity appear at the same time, the human ear can only feel the signal with high intensity, while the signal with low intensity will be masked; the time shadowing effect is when a weak signal is weak. When appearing in a time interval before or after the signal whose intensity is stronger than it, the weaker signal will be obscured, and obviously the concealed signal does not need to be transmitted, thereby reducing the code rate; when the human ear locates the sound source, It is insensitive to the directionality of the low frequency signal. The direction of the high frequency signal is mainly judged from the perception of the envelope, and these characteristics are used to reduce the bit rate of the stereo coding. Of course, this processing loses a part of the original audio signal, but because this part is imperceptible or very insensitive, it basically does not affect the quality of the sound, thereby ensuring the sound quality and reducing the code rate. The compression coding scheme applied in digital audio broadcasting mainly includes Layer II, MPEG AAC in MPEG-1/2, etc., which reduces the bit rate of one channel to 192-96 kbit/s, and may continue to decrease, greatly reducing For the requirements of transmission bandwidth and storage capacity, it should be said that the development of compression coding technology is the basis for the realization of digital audio broadcasting.
2. Network technology
With the development of the broadcasting industry, frequency resources are becoming increasingly tense. In the development of digital audio broadcasting systems, it is necessary to consider saving frequency resources and not contradicting the radio stations that have already worked. In the proposed broadcast system, two methods are mainly used: one is a single frequency network, that is, the entire broadcast network uses the same carrier frequency, and multiple program payloads are on the carrier, and different coverage areas are not required to be allocated differently. Carrier frequency, which saves frequency resources; another method is to use the same carrier rate as the radio station currently in use, depending on the frequency separation and modulation, while broadcasting both analog and digital programs, US IBOC digital audio The broadcasting system is representative of such a scheme.
3. Wireless transmission of high speed data signals
Sound broadcasting can be transmitted through wired channels and wireless channels, but mainly through the propagation of electromagnetic waves in space to transmit signals to receivers, and the propagation characteristics of spatial channels are far less stable than cables, optical fibers, and satellite channels, especially in urban environments. In the case of mobile reception, the frequency selective fading and delay spread due to multipath propagation, as well as the shadow effect caused by building occlusion, severely affect the reception quality, and broadcast systems often operate in such environments. In order to ensure the quality of broadcasting, it is necessary to improve the transmission quality of signals in urban environments and mobile reception situations. The channel coding and modulation sections in digital audio broadcasting systems are mainly designed to solve this problem. At present, most digital audio broadcasting systems adopt error correction coding, orthogonal frequency division multiplexing (OFDM) and a combination of time and frequency interleaving. OFDM is a high frequency spectrum utilization frequency division multiplexing technology, which will convert high speed serial data. Multi-channel parallel data transmission is beneficial for anti-frequency selective fading and delay spread, and it can be implemented with IFFT and FFT to simplify the transmitting and receiving equipment. Experiments show that the transmission system can meet the reliable transmission of broadcast programs under mobile, portable and fixed working conditions.
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