[Home Theater Network HDAV.com.cn] In the audio industry, DSP is a member of the important components of the device, and it is an important symbol of digital audio. We know that DSP does not change the sound in essence, but it enhances the sound effect by tuning, making the music pleasant. DSP is not a product, but a technology. Since it is a technology, it must exist to solve a problem. The DSP exists to process digital audio signals. [1. DSP definition] The full name of DSP is Digital Signal Processor, which is digital signal processing technology. Digital signal processing uses a computer or a dedicated processing device to collect, transform, filter, estimate, enhance, compress, and recognize signals in digital form to obtain a signal form that meets people's needs. A DSP chip refers to a chip capable of implementing digital signal processing technology. The chip's internal Harvard structure with separate programs and data, with a dedicated hardware multiplier, extensive pipeline operation, provides special instructions, can be used to quickly implement a variety of digital signal processing algorithms, real-time running speed up to Tens of thousands of complex instruction programs, far more than general-purpose microprocessors, are increasingly important computer chips in the digital electronics world. The working principle of the DSP is to receive an analog signal and convert it to a digital signal of 0 or 1. The digital signal is modified, deleted, and enhanced, and the digital data is interpreted back into analog data or the actual environment format in other system chips. [two, DSP chip classification] The control system based on the DSP chip is actually a single-chip system, and the various functions required for the entire control can be realized by the DSP chip. Therefore, the volume of the target system can be reduced, the number of external components can be reduced, and the reliability of the system can be increased. For those applications where performance and accuracy are high, real-time is strong, and volume is small, the DSP based chip to form the control system is a very high-performance price-performance method. DSP chips can be classified in the following three ways. According to the basic characteristics - if the DSP chip can work normally at any clock frequency within a certain clock frequency range, such DSP chips are generally called static DSP chips. If there are two or more DSP chips, their instruction set and the corresponding machine code machine pin structure are compatible with each other, then such DSP chip is called a consistent DSP chip. According to the data format, the DSP chip that works in fixed-point format is called fixed-point DSP chip. The floating-point format used by different floating-point DSP chips is not exactly the same. Some DSP chips use a custom floating-point format. The DSP chip uses the IEEE standard floating point format. According to the purpose of the use of - according to the purpose of the DSP, can be divided into general-purpose DSP chips and dedicated DSP chips. [three, the advantages of DSP system] Digital signal processing systems based on general purpose DSP chips have the following advantages over analog signal processing systems: (1) High precision, strong anti-interference ability and good stability. The accuracy is only affected by the quantization error, ie the finite word length, the signal-to-noise ratio is high, and the device performance is small. It is less affected by external factors such as temperature and environment. (2) Easy to program, easy to implement complex algorithms (including adaptive algorithms). The DSP chip provides a high-speed computing platform for complex signal processing. (3) Programmable, no need to redesign, assemble, debug when the system's function and performance change. Such as implementing different digital filtering (low pass, high pass, band pass); radio communication in different working modes in software radio; filters, spectrum analyzers, etc. in virtual instruments. (4) The interface is simple, the electrical characteristics of the system are simple, and the data flow uses a standard protocol. (5) Easy to integrate. (6) Functions that cannot be realized by analog processing: linear phase, multi-sampling rate processing, cascading, easy storage, etc.; (7) Can be used for signals with very low frequencies. The advent of digital signal processors has enabled digital signal processing technology to emerge and develop rapidly. In terms of the quality of audio, digital audio is converted to analog audio quality by analog/digital-to-analog conversion. Digital technology is used in audio editing, synthesis, effect processing, storage, transmission and networking, and in terms of price. Great advantage. Since digital signals are not subject to loss and interference as in analog signals during transmission, more and more audio products use digital input interfaces such as CD players and DVDs. [four, DSP on the audio signal of various applications] Key points include: Active Noise Cancellation, Speech Signal Processing, Audio Processing. Active noise control The traditional passive sound insulation method simply blocks the noise with the soundproof material, and has almost no barrier to the noise generated by the medium and low frequency noise sources. Therefore, it is necessary to use a thick soundproof material to produce an effect. Active noise control is an electronic closed-loop control method that produces a sound that is inverted from the original noise to cancel the original noise (see Figure-1). The advantage is that it is extremely effective in suppressing low frequency noise. The limitation of its application is that it cannot control the noise in the middle and high frequency bands (above 1.5K Hz). In all aspects of communication, annoying noise may occur, and its comprehensive impact is to reduce communication efficiency and success rate. Active noise control technology can improve the signal-to-noise ratio at many levels, and it can dynamically adapt to various conditions compared with traditional simple filters. Uncertain noise that cannot be processed by past filters can be overcome to a considerable extent. Voice signal processing Although many of the data have been digitally encoded, they are sent and received via the original voice communication channel. But voice still holds the first place in all communication content. The demand for the processing of voice signals has increased exponentially in recent years. Speech technology can be divided into the following four items: Speech Enhancement, Speech Recognition, Speech Coding/Decoding, and Echo Suppression. (1) Speech enhancement In the acquisition of voice signals, various pickups (microphones) have different frequency response, directivity, stability, and pick-up mechanism. The microphone combination array with different characteristics can satisfy the user's signal in various frequency bands. A variety of different requirements (Figure 2), the effective grasp of the electroacoustic system achieved under the task of satisfying noise control, so that we can meet the requirements of signal acquisition of various user systems. In signal processing, we can develop different solutions to meet the mission requirements for applications, background noise characteristics, and the relative requirements of speech intelligibility for allowable speech distortion. For example, the requirement of the speech recognition software for the speech signal is different from the requirement of the human ear for the speech signal. Therefore, different procedures are required when completing the communication and when completing the speech recognition task. If the R&D institutions of different missions do not have a comprehensive understanding and grasp of the speech characteristics, it is impossible to achieve truly optimized results on this. In addition, DSP technology performs single-channel signal detection at high speed, multi-channel signal comparison, and its speed can make the user feel that there is no delay in time, and it is completely real-time working effect. (2) Speech recognition The core of the speech recognition system should have the characteristics of less hardware requirements, self-contained time correction, and energy correction. At present, the actual speech is the independent speech recognition of the small vocabulary (200 words) system, and the core of the vocabulary (1800 words) is also completed. In the development direction of automatic speech recognition, it will focus on the development of voice control technology rather than voice input technology. The focus is on the accuracy of the first recognition, rather than the auxiliary recognition of mixed semantics. (3) Voice editing and decoding Due to the powerful functions of the DSP in the speech processing, it is possible to use the "encoded excitation linear prediction" (CELP) algorithm with higher compression in the design and use of speech coding. The open standard currently used is ITU G.723.1. This algorithm is widely used in IP codec and has two transmission rates of 6.3Kbps and 5.3Kbps. It has high voice quality, anti-noise ability and moderate computing load. Can be used by users on a variety of platforms. At the same time, the proprietary 2.4Kbps speech coding algorithm is also under development. It is expected that the algorithm will achieve a better balance between speech quality, anti-noise ability, speech compression ratio, computational load and computation delay. Due to the continuous improvement of hardware performance, it will adapt to the larger computational coding method. According to the principle of information theory, if the high compression ratio is adopted, the relative application is inevitable if the signal index is not lowered. A large amount of codec is used to achieve better audio performance at high compression ratios. (4) Echo suppression In long-distance communication and event communication, it is often troubled by echo. Whether it is a linear echo or an acoustic echo, when the delay exceeds 0.5 seconds, it will be received clearly at the receiving end. For each of these two phenomena, there is a suitable echo suppression algorithm. The DSP-based algorithm is stable and concise, not only suppresses the response speed, but also maintains noise reduction performance for Double Talk, Near-End-Speech and mute state. At the same time, because the linear echo time delay can vary over a wide range of 1 millisecond to 900 milliseconds, there is also a DSP-specific algorithm to overcome the extra load that this variability brings to the system (in a traditional echo suppression system, 300 milliseconds) The delay means that the system performance price ratio is sharply degraded). The source code of these algorithms can also be applied to various communication platforms to solve the problems caused by various aspects of long-distance communication. Music signal processing Since the beginning of the popularization of digital music specifications, due to the elastic factors attached to digital signal processing, many open specifications and proprietary specifications have been produced in the storage, transmission and playback of video and audio signals. For users, the effects they bring, in addition to more durable and cheaper storage media, more diverse receiving channels, also include more beautiful audio-visual effects. However, the audio and video effects obtained by the terminal and the original video source are still expensive and not effective. In order to achieve the so-called "ring field sound effect", various open specifications such as Dolby Surround, Dolby ProLogic, AC-3, and THX are currently available, and commercial decoding chips are also available. But the weakest link in the whole link is in the segment from the speaker system to the human ear. The transfer function of this segment changes randomly due to different listeners and different listening environments, and even varies greatly. The efforts of the original recording engineer are often ruined in this section. And like the traditional audio system, the most difficult part of this performance is often a part of the investment. For this part, DSP proposes a solution. It is independent of the above open specifications to establish an approximate ring-field sound system. In the post-processing stage of the signal, the four-plus-one or five-plus-one channel required by the above specification is simulated with a more humanized two-channel. Requirements and And with DSP dynamic compensation of the variation of the sound field, it is basically possible to use a low-cost DSP-based system to replace the expensive non-DSP high-end system, and completely restore the original recording effect. 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