Dolby Stereo System

The Dolby noise reduction system is a noise reduction circuit based on the application of tape recorders. We know that the poor high-frequency signal-to-noise ratio in the audio band is an inherent defect of tape recorders. The core of Dolby noise reduction technology is "emphasis/de-emphasis", that is, first appropriately increase the treble component of the program, that is, "emphasize" and then record it. As a result, on the tape, the high-frequency signal-to-noise ratio is improved, but for the signal, there is frequency distortion (high-frequency overshoot) at this time. In order to compensate for this distortion, the treble component is appropriately attenuated during playback, that is, "de-emphasis", so that the overshoot part of the program is smoothed. In the process of de-emphasis, the high-frequency noise is attenuated together, so the signal-to-noise ratio in the high-frequency band is improved.
In order to further improve tape recording and playback To improve the signal-to-noise ratio of the machine, Dolby Laboratories has successively proposed three dynamic noise technologies, Dolby ABC, which use the "masking effect" of hearing to determine the value of emphasis/de-emphasis, and the processing methods of different frequency bands are also different. The frequency band division and emphasis values ​​of Dolby ABC are different. Dolby A is mainly used for professional machines, and BC is used for home equipment. Dolby Stereo System The original stereo system consisted of only two channels. Obviously, it is not enough to use two channels to reflect some large-scale performances. For this reason, Dolby (DOLBY) of the United States has carefully designed and invented a 4-2-4 stereo encoding technology ( Figure D is the schematic diagram of this encoder). Its principle It is to express multi-channel stereo programs with four channels, namely left (L), center (C), right (R) and surround (S). However, most audio equipment at that time was mainly two-channel, so in order to make four-channel stereo programs available on two-channel devices, Dolby combined the four channels into two channels through encoding technology. When restoring, only one decoder is needed to restore the two channels to four channels. This is the so-called 4-2-4 encoding technology. The main feature of the Dolby stereo system is the introduction of a real surround sound information (S), which is the difference between it and other simulated surround sound systems. Generally, simulated surround sound has only two basic channels (L, R), and its surround sound is generated through these two basic channels. The post-processing of the channel signal such as phase shift, addition, subtraction, and delay produces a false signal, while Dolby Stereo has a real sense of space and direction. General simulated surround sound can only produce a sense of envelopment. The surround channel (S) of Dolby Stereo is actually a mono signal (some people think it is a dual-channel signal, which is wrong). It does not have the so-called left surround and right surround (except AC-3), but people usually use multiple speakers to create a wide sound field. In fact, the sound emitted by each speaker is the same, but the position is different. Figure D1 is a schematic diagram of the decoder of the Dolby Stereo system. The Dolby Surround Sound System The structure is as follows Figure D2 , from which we can see that the two input signals LT and RT are converted into four signals L', C', R' and S' after entering the decoder. The L' and R' signals are directly taken from the input signals LT and RT. The output signals have the following relationship: L'=L+0.7C+J0.7S J means the same shift of 90 degrees R'=R+0.7C-J0.7S C'=0.7L+0.7R+C S'=0.7L-0.7R+JS From the above four formulas, we can know that each output channel contains the information of other channels. The level of these other channels and the main channel (such as L'=L+0.7C+J0.7S, where 0.7C+J0.7S is the information of other channels) is only 3DB (0.7 times) different. That is to say, the separation between adjacent channels of the four channels is only 3DB. Considering that the center channel C is divided into RT and LT with the same phase and level, the correct intermediate sound image positioning effect can be obtained from L' and R'. In the Dolby surround sound system of Figure D2, the decoder does not process LT.RT, allowing it to be directly output and omitting the intermediate channel to improve the separation effect of the front channel. The separation of the front and rear channels is achieved by a series of processing by the decoder. First, the signal bandwidth is limited to below 7KHZ through a low-pass filter. This measure is mainly to reduce high-frequency crosstalk and prevent the noise generated by the delay circuit from entering the back stage. Then, the improved Dolby noise reducer is added to provide 6DB noise reduction, which means that the ambient noise and crosstalk can be reduced by half at the same time. In summary, the difference between the Dolby surround sound system and the Dolby stereo system is that one less middle channel is used. The Dolby stereo system is suitable for large listening environments such as theaters, while the Dolby surround sound system is suitable for small listening environments such as home theaters. The Dolby directional logic surround sound system The structure is shown in Figure D3 , which is a further improved circuit of Dolby Surround Sound System. There are three differences between Pro Logic Surround and Surround Sound System: 1. A center channel is added, which is consistent with the professional Dolby Stereo System; 2. Adaptive matrix is ​​used to replace the fixed matrix circuit in Dolby Surround Sound; 3. Center mode control is added. The fixed matrix in Dolby Surround Sound System has a single function, which is only used to extract S' (rear surround signal) signal. The adaptive matrix in Pro Logic Surround System will perform very complex functions. It can detect and logarithmically strengthen the dominant signal (that is, strengthen the channel with stronger signal level) according to the strength status of the four channels of LCRS in LT.RT, but keep the volume unchanged, so that the dominant channel direction The sound image positioning is very clear, which is equivalent to improving the separation between signals, so the adaptive matrix circuit is also called the directional enhancement circuit. It increases the separation between channels from the original 3DB to more than 30DB. The Dolby Logic Surround Sound System restores the use of the center channel and sets a center mode controller, which is used to make certain controls on the center channel. In a home theater, people may use low-power middle speakers or medium-power speakers and high-power speakers for their own preferences, or do not use middle speakers for fun. In addition, some families may not use rear surround speakers because the listening environment is small. In order to obtain the best possible environmental effect in all the above situations, it is necessary to select a suitable center mode. Generally, there are three mode settings: 1. Simulation mode: Its function is to distribute the sound of the center channel evenly to the left and right channels for playback. Obviously, this mode is suitable for the case where the center speaker is not used. 2. Normal mode: Its function is to distribute the low-frequency components below 100HZ in the center channel that have little effect on the directionality evenly to the left and right speakers for playback. The center channel plays back the frequency components above 100HZ. This is the most commonly used mode in home theaters (it is suitable for the case of using a low-power center speaker). 3. Broadband mode: In this mode, the system does not distribute the sound of the center channel as above and transmits it as it is. Obviously, it is suitable for the case where a loudspeaker with a larger power is used as the center speaker. In addition, there is usually a "three-channel logic mode" whose function is to send the rear environmental signal to the front channel for playback. Obviously, this mode is suitable for users who do not want to use the rear speakers. In short, the Dolby directional logic surround system uses a directional enhancement circuit to produce a clear sound image in the direction of the speaker. It is particularly effective for single sound source sound such as dialogue environmental effect sound in movies and TV shows. Combining the psychological effects of people's vision and hearing, on the basis of ensuring the front stereo sound field, a good surround effect can be obtained.

AC-3 and Dolby Surround Sound AC-3 System
AC-3 is the audio standard in DVDs. It supports 5.1-channel surround sound, including L (left), R (right), C (center), LS (left rear), RS (right rear) and a 0.1-channel subwoofer. The subwoofer channel has a narrow frequency band (3-120HZ), so it is an auxiliary channel, called 0.1 channel.
AC-3 is a digital encoding method and a flexible audio data compression technology. The most common program source that uses digital encoding is CD records. The digital audio encoding method of CD records is 16-bit PCM, with a sampling frequency of 44.1kHz. Because the amount of data generated by this encoding method is too large, storage and transmission are neither convenient nor economical, and sometimes even impractical. For example, for television broadcasting, the higher the data transmission rate, the greater the bandwidth required for each program. In today's increasingly tight bandwidth resources, too wide a bandwidth is not allowed; for another example, for tangible carriers (laser discs, tapes, etc.), the recording density of each carrier is limited (constrained by the level of technological development at the time), and increasing the amount of data means shortening the length of the program. The capacity of a CD record is about 680MB, which can hold about 1 hour of two-channel PCM digital audio programs. The program capacity will be reduced to about 20 minutes. If it is used to load uncompressed digital video signals, the program capacity will be reduced to tens of seconds, which is of course of no practical value. Therefore, it is necessary to develop a new encoding method that should use less data without causing a significant decrease in the subjective sense of sound quality. This encoding method is called "perceptual coding". It is based on the principle of psychoacoustics and only records those sound signals that can be perceived by human hearing, thereby achieving the purpose of reducing the amount of data while reducing the sound quality.
AC-3 divides the entire audio frequency band into several narrower frequency bands, and the width of each frequency band is not exactly the same, because human hearing has different sensitivities to sounds of different frequencies. Since the useful signal is divided into narrow frequency bands, the problem of filtering down the coding noise is relatively easy, because for each frequency band, all signals outside the frequency band can be completely filtered out without damaging the useful signal. After filtering out the redundant signal, the frequency of the remaining noise signal is very close to the frequency of the useful signal, and then it is filtered out through the masking effect (a psychoacoustic principle: a stronger sound signal can mask the weaker signal in the adjacent frequency band. In other words, if a stronger signal appears in a certain frequency band, then all signals in the frequency band below a certain threshold value will be masked by the strong signal and become inaudible to the human ear. Filtering out these weak signals will not have a negative impact on the sound quality.). It can be seen that the AC-3 coding system is a very effective noise reduction system. These multi-channel digital audio signals divided into narrow frequency bands need to be synthesized into a complete full-band signal in the end, but the amount of data occupied by each frequency band is not evenly distributed. There is an "auditory masking module" inside the encoder, which can simulate the auditory masking effect of human beings. It can determine how the data should be allocated to each frequency band at a certain moment according to the dynamic characteristics of the signal. Sound elements with dense spectrum and loud volume should obtain more data, and those sounds that cannot be heard due to the masking effect should occupy less or no data. Masking module and data allocation technology are key technologies to achieve high efficiency. It can make limited data carry more effective sound signals, which means better sound quality.
From a technical point of view, the dynamic range of AC-3 can reach at least 20bit, the frequency response range is 20Hz-20kHz±0.3dB (-3db at 3Hz and 20.3kHz), and the frequency response range of the bass effect channel is 3-120 Hz±0.3dB (-3dB at 3Hz and 121 Hz). The sampling frequency can be 32kHz, 44kHz or 48kHz, and the bit rate is variable, with a minimum of 32kbit/s (mono mode) and a maximum of 640kbit/s, with typical values ​​of 384 kbit/s (5.1-channel home digital surround sound system) and 192 kbit/s (two-channel stereo system). It can be seen that it can adapt to a variety of different needs.
In audio processing technology, Dolby surround sound is well known. There are three main types of Dolby surround sound systems: one is the ordinary Dolby surround sound system, which has only 3 channels (L, R, S); one is the Dolby directional logic surround sound system, which has 4 channels (L, R, C, S); and one is the Dolby surround sound AC-3 system, which has 6 channels. The first two systems use 2-channel sound equipment to process 4-channel sound matrix coding processing methods, which belong to analog signal processing methods, while the third one uses digital AC-3 compression technology.

Home THX Surround Sound System
Lucas, an American company, launched THX (cinema high-fidelity sound reproduction system) based on the Dolby directional logic decoder. The goal of THX is to make the sound reproduced in the cinema correctly achieve the sound effect of recording studio production. Figure D4 is a block diagram of the THX system. It can be seen that the difference between it and the Dolby system is mainly the application of a unit circuit called "THX Control Center". Through Figure D4, we can clearly see its relationship and connection with the Dolby system. The software used by the THX system is the same as that of the Dolby system, so there must be a Dolby decoder in it. The difference between the two is that the THX system further processes the signal after Dolby decoding, with the purpose of accurately compensating The acoustic characteristics of the space, correcting the timbre imbalance between the main channel and the surround channel, etc., to ensure that the Dolby-encoded sound source program can be reproduced most faithfully. The basic goal of the digital sound field reconstruction system - DSP (three-dimensional stereo system) is to make the sound information heard at home "exactly the same" as what is heard on site. Indoor sound is nothing more than direct sound and reflected sound. However, due to their great differences in time, direction and intensity, the sound heard on site becomes very complex. To reconstruct these sound fields, it is necessary to master the quantitative data of the above sound fields. Only by understanding the quantitative data of each reflected sound can it be simulated by electroacoustic means. First of all, based on According to the shape data of a concert hall and the materials of its various interfaces, a corresponding mathematical model is established, and then the location of the sound source and a certain range of the listening area are determined. The sound line method in geometric acoustics can be used to make the path of each reflected sound of the sound source in the hall, and the direction and intensity of the reflected sound passing through the listening area can be calculated. According to the time interval of each reflected sound, the time series of reflection can be obtained, so as to obtain the characteristic data of the original sound field. Next, the data obtained by computer simulation is used, and the delay amplifier and speaker array are used to use the direct sound signal in the sound source to control the delay amount of each delay device to simulate the direction and intensity of each reflected sound and the order of each reflected sound, thereby reconstructing a similar sound field. So Some advanced AV amplifiers with DSP functions store more than ten kinds of sound field data of different environments, and create more than ten sound field modes by cooperating with Dolby surround system. Note whether this is the input sound source signal or analog signal. The so-called digital means that the analog data of each sound field is stored in the corresponding control chip in digital form by the manufacturer when it leaves the factory. When these data are called, they will control the volume and delay of each auxiliary amplifier to establish the corresponding sound field effect. Through the above introduction, we can know that the principle of DSP is not complicated. Its advantage is that it has no special requirements for the sound source, but its disadvantage is that it cannot reflect the performance environment of the sound source itself.

The heart of home theater-AV amplifierAV
amplifier is the center of home theater. Some analog AV amplifiers on the market, strictly speaking, have no directional function. Only by using Dolby directional logic decoder can people feel a dynamic sound field synchronized with the picture while watching the picture, and produce a sense of presence in a multi-dimensional space. Of course, the audio software used must be marked with DOLBYSURRO UND or THX.
There are three types of directional logic decoders currently sold on the market: Pure decoder, which does not have an amplifier in the machine, but contains a directional logic decoder chip, which can output information of multiple channels, including left, center, right surround (usually two-way) and subwoofer. In this way, users can flexibly choose the amplifier, but there are also requirements for which amplifier to match, otherwise it is difficult to create an ideal sound field effect. From this point of view, the simpler the one, the higher the requirements.
Decoder with center and surround amplifiers, does not have a main amplifier or subwoofer amplifier in the machine, but each channel has a signal output port, including center and surround. In this way, the main amplifier needs to be equipped with an external amplifier, and the subwoofer can be equipped with an amplifier or a source speaker. The center and surround can be equipped with internal amplifiers or external amplifiers.
This model is suitable for families who already have two-channel amplifiers to upgrade their audio systems to home theaters. It is also suitable for audio enthusiasts to upgrade the grade of their equipment. For example, the main channel can be equipped with a higher-grade amplifier such as a tube amplifier to enjoy the melodious and dreamy sound field effect.
Pro-logic decoder with full-channel amplifier. Due to the limitation of the whole machine, the power of the main channel amplifier cannot be made very large. Most of them are made of integrated circuits and can meet the needs of ordinary families. Among them, the power of the main amplifier is about 5OW, the power of the center and surround is 30-4OW, and the power of the subwoofer is above 5OW. AV amplifiers with this decoder are very convenient for users to use. As long as they are equipped with corresponding speakers, plus LD or DVD, a large-screen color TV can form a theater system with a high cost performance.

On Audio Amplifiers
The amplifier in the audio is the key component of the entire audio equipment, so audio enthusiasts are willing to spend manpower, material resources and financial resources to "modify the machine", and constantly improve the power supply, the overall layout of the circuit, the materials used, etc. I am not a super enthusiast, at best I am an audio enthusiast, so here I will talk about my views on audio amplifiers as an audio enthusiast.
Amplifiers are divided into tube amplifiers and transistor amplifiers, let's discuss transistor amplifiers first. The original transistor amplifiers were Class A amplifiers. The working point of the power amplifier tube of this type of amplifier is selected in the linear amplification area of ​​the tube, so even if there is no signal input, the tube has a large current flowing through it, and its load is an output transformer. When the signal is strong, due to the large current, the output transformer is prone to magnetic saturation and distortion. In addition, In order to prevent the tube from entering the nonlinear region, this type of amplifier is often added with a deeper negative feedback, so this power amplifier circuit has low efficiency, small dynamic range, and poor frequency response characteristics. In response to this, people have introduced a Class B push-pull power amplifier. The power amplifier tube of this type of power amplifier circuit works in the Class B state, that is, the working point of the tube is selected in the microchannel pass state, and the two amplifier tubes respectively amplify the positive half-cycle and negative half-cycle of the signal, and then the output is synthesized by the output transformer. Therefore, the currents flowing through the two sets of coils of the output transformer are in opposite directions, which greatly reduces the magnetic saturation phenomenon of the output transformer. In addition, since the tube works in the Class B state, this not only greatly improves the efficiency of the amplifier, but also greatly improves the dynamic range of the amplifier, so that the output power is greatly improved. Therefore, this power amplifier circuit was once popular. .But people soon discovered that this kind of power circuit has the problem of small signal crossover distortion because its power amplifier tube works in Class B working state, and the circuit needs to use two transformers (one output transformer and one input transformer). Since the transformer is an inductive load, the load characteristics are unbalanced in the entire audio band, and the phase shift distortion is serious. For this reason, people have introduced a power amplifier circuit called OTL. This circuit is actually a push-pull circuit, but it removes the two transformers and uses a capacitor and the output load for coupling, which greatly improves the frequency response characteristics of the power amplifier. The power amplifier circuit composed of transistors has made a qualitative leap. Later, people improved this circuit and introduced OCL and BTL circuits. This circuit also uses the output capacitor. Removed, the amplifier and the speaker adopt direct coupling mode, until now the power amplifier circuit composed of transistors, its structure is basically OCL circuit or BTL circuit. The difference between OCL circuit and OTL circuit is that it adopts positive and negative power supply method, so that the output capacitor can be cancelled. BTL circuit is composed of two completely independent power amplifier modules, as shown in Figure C. Part of the signal amplified and output by IC1 passes through the inverting input terminal of IC2, and is amplified and output by IC2 inverting. The load (speaker) is connected between the outputs of the two amplifiers, so that the speaker obtains the composite signal amplified by IC1 and IC2 with a phase difference of 180 degrees.
Whether it is OCL or BTL power amplifier circuit, because it removes the output transformer and output capacitor, the frequency response of the amplifier is widened. In terms of matching with the speaker, when the power amplifier is connected to a speaker with a nominal impedance lower than its rated load impedance, the output power will increase in theory, but this is conditional. The power amplifier must have a small enough output internal resistance and a large enough current gain, and the power supply can provide a large enough working current. Otherwise, not only will the power not increase without distortion, but it will also cause the performance of the amplifier to decline. Another situation is when the power amplifier is connected to a speaker whose nominal impedance is higher than its rated load impedance. In this case, it seems that the power amplifier will be more relaxed, but this is not entirely true. If the power supply voltage capacity of the amplifier is not large enough, voltage overload distortion may occur before the rated output power is reached during playback. In addition, the speaker voice coil will generate induced electromotive force, which has a damping effect on the movement of the speaker. The output impedance of the amplifier has a bypass effect on the induced electromotive force generated by the speaker, thereby effectively suppressing the induced electromotive force of the speaker.

In summary, in order to achieve good sound effects, transistor amplifiers must have lower output impedance, larger current gain, and the power supply must be able to provide sufficiently large working current and higher power supply voltage and have good transient effects.
In order to make the amplifier have lower output impedance and larger current gain, we can use multiple pairs of power tubes in parallel in the rear stage of the amplifier, and select the power amplifier tube with the highest possible withstand voltage value so that it can adapt to loads with different impedances. However, this will increase the driving power. A good amplifier has strict requirements on the power supply. In order to improve the transient response and provide sufficient current, the rectifier tube must use a large current switching rectifier diode (some people call it a high-speed rectifier diode). In addition, the filter capacitor must be more than 10,000 μF. Since the transient current generated by the amplifier during operation is more than 10 amps (depending on the power of the amplifier), the contact resistance and connection resistance of the rear stage cannot be ignored. For example: the circuit has an AC impedance of 0.1 ohms, then at 10 amps Under the action of current, an AC voltage of 1 volt is generated on it. This AC voltage will be coupled to the front stage, which will cause AC interference at the least, and will cause the amplifier to self-excite and damage the power amplifier tube at the worst. We have repaired many high-power power amplifiers, which have burned out the power tubes due to poor contact of the rectifier diode or poor soldering of the filter capacitor. In addition, since high-power power amplifiers have higher gains, the decoupling circuit of the power supply is very important, otherwise it is easy to generate AC noise interference. General power amplifiers require more than two levels of LC filter circuits, and the selection of the grounding point of the filter capacitor is also particular. Finally, there is the power transformer. The overall efficiency of the current power amplifier is about 50%--60%, so the power of the selected power transformer should be selected as follows: The maximum undistorted power of the amplifier/0.5 For example: the power transformer of an amplifier with a maximum undistorted power of 100 watts should have a power of 100/0.5=200 watts. In addition, in order to reduce the interference of the internal resistance and leakage inductance of the power supply to the amplifier, the number of turns per volt should be minimized and the iron core with high magnetic flux rate should be selected in the design of the power transformer. The ring transformer (ring core transformer) is a transformer with better performance.
Here I would also like to mention a very important parameter of the amplifier---dynamic range. We know that now high-end digital audio sources such as CD players and DVD players have adopted high-bit rate digital quantization, and the dynamic range of the output audio source is more than 90db, which is larger than that of traditional recorders (40--70db). Therefore, if the power amplifier does not have enough dynamic range to match it, it is easy to produce peak distortion (clipping effect). The signal waveform of the peak distortion contains extremely rich high-order harmonic components with large power energy. When they are added to the speaker, their energy is very likely to exceed the speaker's bearing power and burn it out. Therefore, in the power amplifier circuit, in order to prevent the amplifier from entering the clipping state, a negative feedback circuit is added to the circuit. Although the negative feedback circuit effectively prevents the generation of clipping, it also causes linear distortion (amplitude distortion) and nonlinear distortion (caused by phase shift) of the signal. Semiconductor device manufacturing has made great progress today, and semiconductor devices with a large dynamic range have been introduced. Under this premise, people have proposed the concept of power amplifiers without negative feedback. Since there is no negative feedback, the fidelity of the amplifier will be further greatly improved.
Now let's talk about the tube amplifier (electronic tube machine) which is sought after by many audio enthusiasts for its soft and pleasant sound quality. It is different from the transistor in the following aspects: 1. The circuit structure of the transistor is more complex than that of the electron tube; 2. The collector current of the transistor is basically not affected by the collector-emitter voltage Vce, while the anode current and anode voltage of the electron tube basically conform to the Ohm's law; 3. The transistor is easily affected by temperature, while temperature has less effect on the electron tube; 4. The transistor works in a low voltage and high current state, so the power supply requirements are high; while the electron tube works in a high voltage and low current state and the power supply requirements are relatively low; 5. The transistor is a current-controlled device with low input and output impedance, while the electron tube is a voltage-controlled device with high input and output impedance, so the electron tube amplifier must have an output transformer to match the load. Due to the electromagnetic inertia of the output transformer and the narrowing of the transmission frequency band (especially the high frequency band), the audio signal is softened and the sound quality sounds soft (in fact, this is not high fidelity); 6. The overload capacity of the electron tube is stronger than that of the transistor, so the dynamic range is relatively higher than that of the transistor, so the sound sounds more pleasant.