| This text studies the Advanced Audio Coding (AAC ) in the source code of systematic letter of Digital Amplitude Modulation Broadcasting (DAMB ). Europe standard DRM can serve long , digital broadcast system , medium-short wave of frequency band at the same time only at present, adopting OFDM system, utilizes signal of the range and phase place at the same time when modulating, it is modulated that is the single sideband, the channel width of 10KHz can be fully utilized , the bandwidth of signal transmitted actually can reach 10KHz , this needs to transform the transmitter, but DAMB here does not change the existing broadcast transmitter, still utilize and take the amplitude modulation system bilaterally. The advantage is that it can realize the signal on a large scale covers , and the receiver is simple , easy to realize , the price is not high. But the question brought is the bandwidth that the reality can be utilized there is half only, the tone quality is influenced (the bandwidth, noise , channel decline), can only be used in channel condition relatively better hitting the digital amplitude modulation broadcasting of wave band. Therefore compared with DRM letter source code, under the same bandwidth, bit rates of DAMB source code output reduces a lot, the compressing rate must raise greatly, the audio coding in its source code adopts AAC .On the basis of input signal, psychological acoustics model rule known already to spend can calculate one estimated value, masking threshold at present (with time related), have used the psychological acoustics model 2 in AAC system. From masking threshold, it can get Noise Masking Rate. In quantization stage , give to any data rate can utilize signal is it quantize the hearing distortedly and minimum of the signal making to shelter definitely, the code carries on Haffman's code in controlling coefficient and quantization coefficient at stage, get the code bit stream.Past audio frequency code among the way, as to initial data, is it reduce samplerate, reduce audio frequency bandwidth , reduce quantization progression traditional method compress the code to use, the compressing rate can probably be up to 15:1. But too much use of these methods can make apprehending and discerning degree of perceptible audio quality drop too much. AAC uses the psychological acoustics model, regards perceptible audio quality as the standard and compresses the code, on the premise of guaranteeing perceptible audio quality, can reach higher compression ratio.This text code improved optimization software realize of AAC. It include control to audio frequency bandwidth among them , psychological acoustics model algorithm and quantize code module ,etc. In order to be suitable for the transmission in the digital amplitude modulation broadcasting channel, the frame length of AAC code has changed , having use the correspondingfast algorithm, and the new tool that the error protected joined.Using Visual C ++ tool under the operating system Windows XP, it reduce bit rate of MPEG-4 AAC to10/16kbps that this system can supported to. After decoding the bandwidth of the audio frequency is 50Hz-7kHz that correspond with AM radio sets, which can keep the tone quality as good as possible. On the basis of t the theory analysis, it has adjusted the parameter of MPEG-4 AAC, in order to adapt to the bandwidth of audio frequency after changing bandwidth of the audio frequency and bit rate. |
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