Dolby Stereo System

Dolby noise reduction system db-1.gif (438 bytes)
The Dolby noise reduction system is a noise reduction circuit based on the application of tape drives. We know that the poor high-frequency signal-to-noise ratio of the audio section is a congenital defect of the tape recorder. The core of Dolby noise reduction technology is "emphasis / de-emphasis". First increase the treble component of the program appropriately, that is, "emphasize" and then record, so on the tape, the high-frequency signal-to-noise ratio is increased, but for the signal, there is frequency distortion (high frequency overshoot) In order to compensate for this distortion, the treble component is appropriately attenuated during playback, that is, "de-emphasized", so that the overshoot part of the program will be restored. In the process of de-emphasis, high-frequency noise is attenuated together, so the high frequency band The signal-to-noise ratio is improved.
In order to further improve the signal-to-noise ratio of the tape recorder, Dolby Laboratories has proposed three dynamic noise technologies of Dolby ABC. These technologies use the auditory "masking effect" to determine the amount of emphasis / de-emphasis, and different frequency bands The processing methods are also different.The frequency division of Dolby ABC is different from each other.Dolby A is mainly used for professional machines, and BC is used for household equipment.

Dolby Stereo System db1-1.gif (539 bytes)
The original stereo system consisted of only two channels. Obviously, it is not enough to use two channels to reflect some large-scale performance music. For this reason, Dolby Corporation of the United States has invented a 4 -2-4 stereo coding technology (Figure D is the principle diagram of this encoder), its principle is to surround the multi-channel stereo program with four channels, that is, left (L) middle (C) right (R) surround (S) to show. But most of the audio equipment at that time was mainly two-channel, so in order to make four-channel stereo programs can be used on two-channel equipment, Dolby uses encoding technology to convert four channels Combined into two channels, only one decoder is needed to restore the two channels to four channels during the restoration. 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), this is the difference between it and other analog surround sound systems. Generally, analog surround sound has only two basic channels (L, R), and its surround sound is shifted through these two basic channel signals. Phase, addition, subtraction, delay and other post-processing produce a false signal, while Dolby stereo has a real sense of space and orientation. General analog surround sound can only produce a sense of surround. The surround channel (S) of Dolby stereo is actually a mono signal (some people think it is a two-channel signal, which is wrong), it is not There are so-called left surround and right surround (except AC-3), but people usually use multiple speakers to create a wide sound field from the current surround channel, in fact, the sound emitted by each speaker is the same, It's just a different position. Figure D1 is a schematic diagram of the decoder of the Dolby stereo system.

Dolby Surround Sound System db2-1.gif (567 bytes)
Its structure is shown in Figure D2. It can be seen from the figure that the two input signals LT.RT form a decoder and form L'.C'.R'.S 'four channels. .The output signals have the following relationship:
L '= L + 0.7C + J0.7S J means shift the same 90 degrees
R '= R + 0.7C-J0.7S
C '= 0.7L + 0.7R + C
S '= 0.7L-0.7R + JS
As can be seen from the above four types, each output channel contains information of other channels, the information of these other channels and the main channel (such as L '= L + 0.7C + J0.7S, of which 0.7C + (J0.7S is the information of other channels) The level differs only by 3DB (0.7 times), which means that the separation between the adjacent channels of the four channels is only 3DB, considering that the center channel C is the same phase and electric It is equally divided into RT and LT, so the correct intermediate sound image localization effect can be obtained from L 'and R'. In the Dolby surround sound system in Figure D2, the decoder does not perform any processing on LT.RT, so that Direct output without the middle channel to improve the separation effect of the front channel. The separation of the channels placed in the front and back directions is achieved by the decoder performing a series of processing on the rear, first delay and then through a low-pass filter The device limits the signal bandwidth to below 7KHZ.This measure is mainly to reduce high-frequency crosstalk, while preventing the noise generated by the delay circuit from entering the post stage, and then adding an improved Dolby noise reducer to provide a 6DB reduction. The amount of noise, that is to say, can reduce environmental noise and crosstalk by half at the same time.
In summary, the difference between the Dolby surround sound system and the Dolby stereo system is that a middle channel is used less.The Dolby stereo system is suitable for listening environments in large scenes such as theaters, while the Dolby surround sound system It is suitable for small scene listening environment such as home theater.

Dolby Pro Logic Surround Sound System DB3-1.gif (686 bytes)
Its structure is shown in Figure D3, which is a circuit that further improves the Dolby surround sound system. There are three differences between the directional logic surround and the surround sound system: 1. An additional center channel is added, which is consistent with the professional Dolby stereo system ; 2. The adaptive matrix is ​​used to replace the fixed matrix circuit in Dolby Surround; 3. The center mode control is added.
The fixed matrix function in the Dolby surround sound system has a single function, just to take out the S '(back surround signal) signal. The adaptive matrix in the directional logic surround system will perform a very complex function, which can be based on the LT.RT The signal strength of the four channels of the LCRS is detected and the dominant signal is strengthened logarithmically (that is, the channel with the stronger signal level is strengthened), but the volume can be kept unchanged, so that the direction of the dominant channel The sound image localization on the above is very clear, which is equivalent to improving the separation between the signals, so the adaptive matrix circuit is also called the direction enhancement circuit.It increases the separation between the channels from the original 3DB to more than 30DB.
The Dolby Logic surround sound system restores the use of the center channel and sets the center mode controller. Its function is to control the center channel. In home theaters, people may use low power for their own preferences. The middle speaker or the middle power speaker and the high power speaker or dry fun do not use the middle speaker.In addition, some homes may not use the surround back speaker because the listening environment is small.In order to make the above situations as possible as possible For good environmental effects, you need to select the appropriate center mode.There are generally three mode settings:
1. Emulation mode: Its role is to distribute the sound of the center channel to the left and right channels for playback evenly. Obviously this mode is suitable for the case where the middle speaker is not used.
2. Normal mode: Its function is to evenly distribute the low-frequency components below 100HZ in the center channel that have little effect on the directionality to the left and right speakers for playback. The center channel replays the frequency components higher than 100HZ. This is the most commonly used mode in home theaters (it is suitable for small power center speakers).
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.It is obviously suitable for the case of using a higher-power speaker as the center speaker.
In addition, there is usually a "three-channel logic mode" whose role 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 the sound of a single sound source such as the dialogue environment effect sound in the film and television film. The psychological effect of hearing can obtain a good surround effect on the basis of ensuring the front stereo sound field.

AC-3 and Dolby Surround AC-3 system
AC-3 is the audio standard in DVD, which supports 5.1-channel surround sound, including L (left), R (right), C (center), LS (rear left), RS (rear right) and a 0.1 The sub-bass of the channel, this sub-bass 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 other most common source of programs now using digital encoding is CD records. The digital audio coding method of CD records is 16bit PCM, and the sampling frequency is 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 impossible. For example, for TV broadcasting, the higher the data transmission rate, the greater the bandwidth required for each set of programs. Today, with the increasing frequency band resources, excessively wide frequency bands are not allowed; for example: for tangible carriers ( (Laser discs, magnetic tapes, etc.), the recording density of each carrier is limited (restricted by the degree of technological development at the time), 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 program, the program capacity will be reduced to about 20 minutes, if used to load digital video signals without any compression, the program capacity will be reduced To tens of seconds, this of course has no practical value. Therefore, it is necessary to develop a new encoding method, which should use less data volume without causing a significant decrease in the subjective listening perception of sound quality. This encoding method is called "Perceptual Coding". It is based on the principle of psychoacoustics and records only those sound signals that can be perceived by human hearing, thereby reducing the amount of data and reducing the sound quality. .
AC-3 divides the entire audio frequency band into several narrower frequency bands, the width of each frequency band is not exactly the same, because human hearing has different sensitivity to sounds of different frequencies. Since the useful signal is divided into narrow frequency bands, the coding noise filtering problem is relatively easy, because for each frequency band, all signals outside the frequency band can be filtered out without damaging the useful signals. After filtering out the excess signal, the frequency of the remaining noise signal is very close to the frequency of the useful signal, and then 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 it is in a certain frequency band A strong signal appears in the channel, then all signals in the band below a certain threshold will be masked by the strong signal and become inaudible to the human ear. Filtering these weak signals will not cause bad sound quality. Impact.) Filter it out. It can be seen that the AC-3 encoding system is a very effective noise reduction system. These multi-channel digital audio signals divided into narrow frequency bands still need to be synthesized into a complete full-band signal, but the amount of data occupied by each frequency band is not evenly distributed. There is a "hearing masking block" inside the encoder It can simulate the human hearing masking effect, which can be determined according to the dynamic characteristics of the signal: at a certain moment, how the data volume should be allocated to each frequency band is the most appropriate. Sound elements with dense spectrum and high volume should get more data occupancy, and those sounds that are not heard due to the masking effect will occupy less or no data. The masking module and the data volume distribution technology are the key technologies to obtain high efficiency. It can make the limited data volume 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). Sampling frequency can be 32kHz, 44kHz or 48kHz, bit rate is variable, the lowest is 32kbit / s (mono mode), the highest is 640kbit / s, the typical value is 384 kbit / s (5.1 channel home digital surround sound System) and 192 kbit / s (two-channel stereo system). It can be seen from this that it can adapt to many different needs.
In audio processing technology, Dolby surround sound is well known. There are three main types of Dolby surround sound system. One is the ordinary Dolby surround sound system, with only 3 channels (L, R, S); one One is Dolby Directional Logic Surround Sound System with 4 channels (L, R, C, S); one is Dolby Surround Sound AC-3 system with 6 channels. The first two systems use 2-channel sound equipment to process 4-channel sound matrix coding processing, which belongs to the analog signal processing method, and the third type uses digital AC-3 compression technology.

Home THX surround sound systemU.S. Lukas company launched THX (cinema high-fidelity audio playback system) on the basis of Dolby directional logic decoder.The goal of THX is to make the playback sound in the theater correctly reach the studio production. Acoustic effects. Figure D4 is a block diagram of a 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 that it and Du The relationship and connection of the Than system, the THX system uses the same software as the Dolby system, so there must be a Dolby decoder, the difference between the two is that the THX system performs further processing on the Dolby decoded signal, the purpose is Accurately compensate the acoustic characteristics of the space, correct the imbalance in the timbre between the main channel and the surround channel, etc., so as to ensure the most faithful playback of Dolby encoded sound source programs.

Digital sound field reconstruction system-DSP (three-dimensional stereo system)
The basic goal of this system is to make the sound information that people hear under the conditions of the home be "the same as" heard on the spot. The indoor sound is nothing more than composed of direct sound and reflected sound, but due to their time, direction and intensity, etc. The sounds heard at the scene become very complicated, and it is necessary to master the quantitative data of the above sound field to reconstruct these sound fields. As long as you understand the quantitative data of each reflected sound, you can simulate it by electroacoustic means, first based on a certain music The shape data of the hall and its various interface materials establish a corresponding mathematical model, and then determine the sound source position and a certain range of listening area, and use the sound line method in geometric acoustics to make the path of each reflected sound of the sound source in the hall , The direction and intensity of the reflected sound through the listening area can be obtained, and the time series of reflection can be obtained according to the time interval of each reflected sound, so that the characteristic data of the original sound field can be obtained. Next, the data obtained by computer simulation, Use delay amplifier and speaker array to simulate each reflection by using the direct sound signal from the sound source to control the delay of each delay The sound direction and intensity are the same as the reflected sound to reconstruct a similar sound field. Therefore, some advanced AV amplifiers with DSP function store sound field data of more than ten different environments, and create more than ten by working with Dolby surround system. A sound field mode. Note whether this is the input sound source signal or an analog signal. The so-called digital means that the analog data of each sound field is stored by the manufacturer in the digital form on the corresponding control chip when the factory leaves the factory. Data, these data will control the volume and delay of each auxiliary amplifier to establish the corresponding sound field effect.
Through the above introduction, we know that the principle of DSP is not complicated. Its advantage is that it has no special requirements for the sound source, but its shortcoming is that it cannot reflect the playing environment of the sound source itself.

The heart of the home theater ---- AV power amplifier AV power amplifier is the center of the home theater. There are some analog AV power amplifiers on the market, strictly speaking, there is no directional function. Only by using Dolby directional logic decoder can people really feel a synchronization with the picture while watching the picture. The dynamic sound field produces 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 on the market: pure decoders, without amplifiers, but with directional decoder chips, which can output multiple channels of information, including left. Center and right surround (usually two-way) heavy bass, etc. In this way, the user can flexibly choose and match the power amplifier, but also pay attention to which power amplifier is equipped, otherwise it is difficult to create an ideal sound field effect. From this point of view, the simpler the more demanding.
The decoder with center and surround amplifiers does not have the main amplifier and subwoofer amplifier, but each channel has a signal output port, including center and surround. In this way, the main power amplifier should be equipped with an external power amplifier. The subwoofer can be equipped with a power amplifier or a source speaker. You can choose the built-in power amplifier for center and surround, or it can be equipped externally.
This model is suitable for families who originally had two-channel power amplifiers, and upgraded their audio systems to home theaters. It is also suitable for audio enthusiasts to upgrade the grade of equipment. For example, the main channel can be equipped with a higher-end amplifier such as a tube amplifier to enjoy the melodious and dreamy sound field effect.
Directional logic decoder with full-channel power amplifier. Due to the limitation of the whole machine, the power of its main channel power amplifier cannot be made very large. Most of them are made of integrated circuits, which can meet the needs of ordinary families. Among them, the power of the main power amplifier is about 5OW, the center and surround power are 3O-4OW, and the subwoofer power is above 5OW. The AV power amplifier with this decoder is very convenient for users to use. As long as it is equipped with the corresponding speakers, plus LD or DVD, a large-screen color TV can form a theater system, which has a high cost performance.

On the audio amplifier, the power amplifier in the audio is a key part of the entire audio equipment, so audio enthusiasts spend their time and money on the "motorcycle" on the power supply, the overall layout of the circuit, the materials, etc. Continuous improvement. I am not a super audiophile, at best I am an audio enthusiast, so I will talk about my views on audio amplifiers as an audio enthusiast here.
The power amplifier is divided into bile machine and stone machine, first discuss the stone machine. The initial power amplifier of the stone machine is a class A amplifier. The working point of the power amplifier tube of this type of power amplifier is selected in the linear amplification area of ​​the tube, so even if there is no signal input, The tube also has a large current flowing through it, and its load is an output transformer.When the signal is strong, the output transformer is prone to magnetic saturation and distortion due to the large current.In addition, in order to prevent the tube from entering the nonlinear region, such amplifiers are often all Deeper negative feedback is added, so this power amplifier circuit has low efficiency, small dynamic range, and poor frequency response characteristics. For this, people have introduced a class B push-pull power amplifier. This type of power amplifier circuit works with its power amplifier tube. In the class B state, that is, the working point of the tube is selected in the micro channel pass state, the two amplifier tubes respectively amplify the positive half cycle and the negative half cycle of the signal, and then the output transformer synthesizes the output. Therefore, the two coils flowing through the output transformer have opposite current 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 amplifier The dynamic range of the amplifier greatly improves the output power. So this power amplifier circuit was popular for a while. But people soon discovered that this type of power circuit has small signal crossover distortion due to its power amplifier tube working in Class B working state. Problem, and the circuit needs to use two transformers (one output transformer and one input transformer), because the transformer is an inductive load, so the load characteristics are unbalanced and the phase shift distortion is more serious in the entire audio section. A kind of power amplifier circuit called OTL. The form of this circuit is actually a form of push-pull circuit, except that two transformers are removed, and a capacitor is used to couple with the output load, which greatly improves the power amplifier. The frequency response characteristics of the transistor. The power amplifier circuit composed of transistors has made a qualitative leap.Later, people have improved this circuit and introduced OCL and BTL circuits. This circuit also removes the output capacitor, and the amplifier and the speaker adopt a direct coupling method. Until now, the power amplifier circuit composed of transistors is basically an OCL circuit or a BTL circuit. The difference between the OCL circuit and the OTL circuit is that a positive and negative power supply is adopted. The electrical method can eliminate the output capacitor. The BTL circuit is composed of two completely independent power amplifier modules, as shown in Figure C. A part of the signal output by IC1 is passed through the inverting input of IC2, and the output is inverted by IC2. , The load (speaker) is connected between the outputs of the two amplifiers, so that the speaker obtains a composite signal amplified by IC1 and IC2 with a phase difference of 180 degrees. yx1.gif (1719 bytes)
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 broadened. In connection with the speaker, when the power amplifier is connected to a speaker whose nominal impedance is lower than its rated load impedance, the theoretical output power will increase, but this is conditional, the power amplifier must have a sufficiently small output internal resistance and must With a sufficiently large current gain, the power supply can provide a large enough operating current, otherwise the power will not increase without distortion but will cause the performance of the amplifier to decline. Another situation is that the power amplifier is connected to a speaker whose nominal impedance is higher than its rated load impedance.At this time, it seems that the power amplifier will be easier, but in fact it is not the case.If the power supply voltage capacity of the amplifier is not large enough, it may not reach it during playback. Voltage overload distortion occurs at the rated output power. In addition, the speaker voice coil will generate an induced electromotive force.This induced electromotive force 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, which can effectively suppress the induced electromotive force of the speaker. In summary, in order to obtain a good sound effect, the transistor power amplifier must have a lower output impedance, a larger current gain, and the power supply must provide a sufficiently large operating current and a higher power supply voltage and transient effects. it is good.
In order to make the amplifier have a lower output impedance and a larger current gain, we can use multiple pairs of power tubes in parallel to achieve the final stage of the power amplifier, and select the power amplifier tube with the highest voltage resistance as possible, so that it can adapt to different impedance loads But this move will increase the driving power. A good power amplifier has strict requirements on the power supply. In order to improve the transient response and provide enough current rectifier, a large current switching rectifier diode (some people call it high speed) (Rectifier diode), in addition, the filter capacitor should be more than 10,000μF. Since the transient current generated by the power amplifier during operation reaches more than 10A (depending on the power of the power amplifier), the contact resistance and wiring resistance of the subsequent stage can not be ignored For example, if the circuit has an AC impedance of 0.1 ohms, then an AC voltage of 1 volt will be generated thereon under the effect of a current of 10 amps.This AC voltage will be coupled to the pre-stage. The amplifier is self-excited and damages the power amplifier tube. We have repaired many high-power amplifiers, which caused burnout of the power tube due to poor contact of the rectifier diode or virtual welding of the filter capacitor. In addition, due to the high-power amplifier All the amplifiers have high gain, so the decoupling circuit of the power supply is very important, otherwise it is easy to produce hum interference. General power amplifiers require two or more LC filter circuits, and the selection of the ground point of the filter capacitor is Pay attention. 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: the maximum undistorted power of the amplifier / 0.5 For example: one For a power amplifier with a maximum undistorted power of 100 watts, the power of the power transformer should be 100 / 0.5 = 200 watts. In addition, in order to reduce the interference of the internal resistance and leakage inductance of the power supply on the amplifier, the design of the power transformer should minimize each volt turn. The number and selection of iron cores with high magnetic flux rate. Ring cattle (ring iron core transformer) is a transformer with better performance.
Here I also want to mention that a very important parameter of the power amplifier is the dynamic range. We know that high-end digital audio sources such as CD players and DVD players have adopted high bit rate digital quantization. The dynamic range of the sound source is more than 90db than the traditional recorder (40-70db). Therefore, if the power amplifier does not have enough dynamic range to match it, it is easy to produce peak-cut distortion (clipping effect). The waveform contains very rich high-harmonic components with great power and energy. When they are added to the speaker, their energy is likely to exceed the speaker's withstand power and cause it to burn 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. The manufacture of semiconductor devices has made great progress today, and semiconductor devices with large dynamic ranges 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 It has further been greatly improved.
Now let's talk about the amplifier (tube machine), which is sought after by many audio enthusiasts for its soft and pleasant sound quality. It differs from transistors in the following aspects: 1. The circuit structure of the transistor is more complicated than the tube; 2. The collector current of the transistor is basically not affected by the collector-emitter voltage Vce, and the anode current and anode voltage of the tube Conform to the law of European mothers; 3. Transistors are susceptible to temperature, but the temperature has little effect on the tube; 4. Transistors work in a low voltage and high current state, so the requirements for power supply are high; The requirements of the power supply are relatively low; 5. The transistor is a current control device with low input and output impedance, and the tube is a voltage control device with high input and output impedance. Therefore, the 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 is soft (in fact, this is not high fidelity); So the dynamic range is relatively higher than the transistor, so the sound sounds more pleasant.

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