When you are enjoying wonderful high-quality music, have you noticed that the two major components of restoring high-quality music are: high-quality video sources (software) and high-fidelity equipment (hardware)? The role of video sources and equipment in the process of restoring music is the same as the relationship between computer software and hardware. Neither can be separated from the other.
In ancient times, when people heard good music, once the performance ended, the sound disappeared forever, and they could only sigh, "This music should only exist in heaven, how many times can it be heard on earth?" However, since Edison invented the phonograph in 1877, electroacoustic technology has developed rapidly in the past 100 years. From coarse-groove records to micro-groove LP records, and then to digital laser discs CDs, laser video discs LDs, and today's VCDs and DVDs, there are also magnetic recording devices such as recorders, video recorders, and digital magnetic recording devices, etc. Others such as amplifiers and speaker manufacturing have made rapid progress in terms of principles and production.
Thanks to the advancement of electronic and electroacoustic technology, people can not only record beautiful and pleasant sounds, but also record them as realistically as possible and replay them as realistically as possible. This kind of audio equipment is generally called high-fidelity audio (Hi-Fi, High fidelity). Fever audio equipment is basically the same as high-fidelity audio equipment.
Compared with predecessors, current music lovers are fortunate. As long as you own a set of high-fidelity audio equipment, you can enjoy all kinds of music you like at any time without leaving home. You can also enjoy the wonderful performances of world-class orchestras conducted by famous conductors, and the wonderful performances of famous performers, and listen to them repeatedly at any time.
In Beijing, Shanghai, Guangzhou, Hong Kong, Taiwan, and almost all economically developed countries and regions, there are a large number of audio enthusiasts. Now there is a popular term, audio enthusiasts, or enthusiasts for short. The corresponding high-fidelity audio (Hi-Fi) also has an enduring market and has become an important part of the consumer electronics market.
The fever for audio equipment has promoted the development of audio equipment and the improvement of cultural taste. Therefore, the fever activities are welcomed by the public. As for the excessive preference of audio enthusiasts for music and audio equipment, the public also adopts a tolerant attitude. As for the very few people who are obsessed with audio equipment, they only need to be guided appropriately. Some people buy audio equipment to show off their wealth and satisfy their vanity by importing expensive audio equipment. This has nothing to do with music and audio enthusiasts.
In the future, I will introduce to you some basic knowledge about audio equipment, knowledge on equipment selection and matching, knowledge on DIY and some other related issues. Due to my limited personal level, most of the things are reproduced from the Internet or modified based on the original text, which involves the understanding of some specific issues or specific products, so there will inevitably be some bias. I hope you understand and welcome everyone to discuss.
1. Explanation of terms and terminology of audio knowledge.
1. The development history of audio technology.
The development history of audio technology can be divided into four stages: electron tubes, transistors, integrated circuits, and field effect tubes. In 1906, American De Forest invented the vacuum triode, which pioneered the electroacoustic technology of mankind. After the negative feedback technology was invented by Bell Labs in 1927, the development of audio technology entered a new era. The more representative one, such as the "Williamson" amplifier, successfully applied the negative feedback technology to greatly reduce the distortion of the amplifier. By the 1950s, the development of electron tube amplifiers reached a climax, and various electron tube amplifiers emerged in an endless stream. Because the sound of electron tube amplifiers is sweet and mellow, it is still preferred by audiophiles. The emergence of transistors in the 1960s brought audio enthusiasts into a wider world of audio. Transistor amplifiers have the characteristics of delicate and moving sound, low distortion, wide frequency response and dynamic range. In the early 1960s, the United States first introduced a new member of audio technology-integrated circuits. In the early 1970s, integrated circuits were gradually recognized by the audio industry for their high quality, low price, small size, and multiple functions. Up to now, thick film audio integrated circuits and operational amplifier integrated circuits have been widely used in audio circuits. In the mid-1970s, Japan produced the first field effect power tube. Because field effect power tubes have the pure and sweet sound of electron tubes, as well as the characteristics of a dynamic range of 90dB and total harmonic distortion THD <0.01% (at 100kHz), they quickly became popular in the audio industry. Many amplifiers today use field effect tubes as the final output. The development of audio technology has gone through the historical periods of electron tubes, transistors, and field effect tubes, and each has its own characteristics in different historical periods. It is expected that the mainstream of audio technology in the future will be digital audio technology.
Let me introduce the specific meaning of dB. The unit dB is widely used in electronics. It is a relative unit for measuring and comparing the power, voltage and current of a system. Later, due to the advancement of science and technology, it was realized that human response to sound changes according to the logarithmic law, so a unit was created, which is Bel, the name of the inventor of the telephone. Its expression is: Bel=lg(P/Po)P is the power to be measured Po is the reference power: Bel represents the logarithm with base 10. In practice, it is found that Bel is too large, so one tenth of it is taken as a new unit, that is, decibel (dB). Dividing Bel by 10 is dB. The expression is: dB=10lg(P/Po), dB=20lg(E/Eo), dB=20lg(I/Io).
2. What is Hi-Fi?
What kind of audio equipment is Hi-Fi? Hi-Fi is the abbreviation of High-Fidelity in English, which literally means "high fidelity". Its definition is: the playback sound is highly similar to the original sound. So what kind of audio equipment can play back the sound of Hi-Fi? So far, it is still difficult to make a definite conclusion. Professionals in the audio industry use various instruments and various means to detect various indicators to determine the degree of Hi-Fi of the equipment, while audio enthusiasts often use their own ears to judge whether the equipment has reached the Hi-Fi in their minds. To determine the high fidelity of the playback sound, not only high-performance equipment and software are required, but also a good listening environment. Therefore, how to correctly measure the Hi-Fi degree of audio equipment still has the difference between objective testing and subjective evaluation.
3. Main technical indicators of the sound system.
The quality of the overall technical performance of the sound system depends on the performance of each unit. If the technical indicators of each unit in the system are high, the overall technical indicators of the system are very good. There are six main technical indicators: frequency response, signal-to-noise ratio, dynamic range, distortion, transient response, stereo separation, and stereo balance.
1. Frequency response: The so-called frequency response refers to the frequency range of the audio equipment when it is playing back and the relationship between the amplitude of the sound wave and the frequency. Generally, the frequency amplitude of 1000Hz is used as a reference for testing this indicator, and the frequency amplitude is expressed in decibels (dB) in logarithmic units. The overall frequency response of the audio system is theoretically required to be 20~20000Hz. In actual use, due to reasons such as circuit structure and component quality, this requirement is often not met, but it is generally required to reach at least 32~18000Hz.
2. Signal-to-noise ratio:
The so-called signal-to-noise ratio refers to the ratio of the sound system's playback of the sound source software to the new noise generated by the entire system. The noise mainly includes thermal noise, AC noise, mechanical noise, etc. This indicator is generally tested in decibels (dB) of the logarithmic ratio of the rated output power of the playback signal to the system noise output power when there is no signal input. The signal-to-noise ratio of a general sound system must be above 85dB.
3. Dynamic range:
Dynamic range refers to the logarithmic value of the ratio of the maximum undistorted output power of the sound system during playback to the system noise output power during static state, expressed in decibels (dB). Generally, the dynamic range of a sound system with good performance is above 100 (dB).
4. Distortion: Distortion refers to the change of some parts (waveform, frequency, etc.) of the original sound source signal after the sound system replays the sound source signal. There are mainly the following types of distortion in the sound system: 1. Harmonic distortion: The so-called harmonic distortion refers to the sound after the sound system replays has many additional harmonic components compared to the original signal source. This additional harmonic component signal is the multiple or frequency division of the signal source frequency, which is caused by the nonlinear characteristics of the negative feedback network or amplifier. The harmonic distortion of the high-fidelity sound system should be less than 1%. 2. Intermodulation distortion: Intermodulation distortion is also a kind of nonlinear distortion. It is a mixture of two or more frequency components in a certain proportion. The frequency signals are modulated with each other. After passing through the sound playback equipment, a new nonlinear signal is generated. The signal includes the sum and difference signals between the various signals. 3. Transient distortion: Transient distortion is also called transient response. It is mainly caused by the slow response of the amplifier when a large transient signal is suddenly added to the amplifier, which causes the signal to be distorted. Generally, the ability of an amplifier to follow transient signals is expressed by observing whether the envelope waveform of the amplifier output signal is similar to the input square wave waveform after the input square wave signal passes through the playback equipment.
5. Stereo separation: Stereo separation refers to the degree of isolation between the left and right channels in a stereo sound system. It actually reflects the degree of crosstalk between the left and right channels. If the crosstalk between the two channels is large, the stereoscopic sense of the reproduced sound will be weakened.
6. Stereo balance:
Stereo balance refers to the difference in gain between the left and right channels in a stereo playback system. If the imbalance is too large, the stereo image positioning of the reproduced stereo will be offset. Generally, the stereo balance of a high-quality audio system should be less than 1dB.
4. How is the range of sound and the audio frequency range of the sound reproduced by the sound system divided? How does each frequency band perform for music?
The range of the sound reproduced by the sound system can generally be divided into eight ranges: sub-bass, bass, mid-bass, mid-range, mid-treble, sub-treble, treble, and ultra-treble. The audio frequency range can generally be divided into four frequency bands, namely low frequency band (30~150Hz); mid-range frequency band (150~500Hz); mid-high frequency band (500~5000Hz); high frequency band (5000~20000Hz). Among them, the 30~150Hz frequency band: can express the low-frequency components of music, allowing the audience to feel a strong and powerful dynamic. The 150~500Hz frequency band: can express the expressiveness of a single percussion instrument in music, and is the part of the low frequency that expresses strength. The 500~5000Hz frequency band: mainly expresses the clarity of the singer's language and the expressiveness of string music. The 5000~20000Hz frequency band: mainly expresses the brightness of music, but too much will make the sound broken.
5. What are the common terms used by audiophiles?
The terms commonly used by audiophiles are relatively abstract. The commonly used terms are as follows: 1. Nerve line: mainly refers to the signal line that transmits low-level (millivolt, microvolt level) and small current. Generally, the nerve line is used for both audio and video. The plugs at both ends of the higher-level nerve line are gold-plated RCA plugs, and the surface of the wire is coated with an anti-static protective layer. 2. Fever line: mainly refers to the speaker signal transmission line with a larger cross-section and a larger number of strands. High-quality fever lines are made of materials such as oxygen-free copper. 3. Burn-in: The so-called burn-in is similar to the running-in period of mechanical machines. That is, after the audio equipment has worked for a certain period of time, the temperature inside the machine is the same as the ambient temperature, so that the working state of each level of amplifier reaches the best point, and the sound reproduced at this time is the best. 4. Modify: The so-called modify comes from the English word Modify, which means to correct and modify. Enthusiasts replace and modify the components or circuits in the audio system to upgrade it, which is called modifying. 5. Explosion: The so-called explosion refers to the deafening atmosphere produced when the audio equipment is replaying and the music reaches its climax. 6. Tube amps: Tube amps refer to amplifiers made with electron tubes. The warm and transparent sound quality of electron tube amplifiers is still unforgettable to the older generation of audiophiles. 7. Transistor amps: Transistor amps refer to amplifiers made with transistors. 8. Tube amps: Audio equipment made with a mixture of electron tubes and transistors. Generally, electron tubes are used as preamplifiers and transistors are used as postamplifiers. 9. Ring transformers: Ring transformers refer to toroidal transformers, which have less magnetic leakage than ordinary transformers. 10. Large ponds: Large ponds refer to power filter capacitors, which are generally large-capacity capacitors of more than 10,000 μF. 11. Beautiful sound: refers to the high-quality sound reproduction of audio equipment, which meets the requirements of high fidelity. 12. Resolution: refers to the transparency of the sound reproduction of audio equipment, giving people a feeling of "clear bottom". 13. Coloration: Coloration refers to the sound that is not in the reproduced sound caused by the resonance of other objects or materials due to the vibration of sound waves during the reproduction process. It is harmful to the effect of reproduction. 14. Microphone: refers to various microphones. 15. Tonic: refers to high-quality components used when modifying the sound system.
6. How should the speakers be placed?
The correct placement of the speakers is one of the factors to achieve a good sound effect. When placing them, you must pay attention to the following issues: 1. The distance between the two speakers should not be less than 1.5~2 meters, and they should be kept at the same level. The distance between the left and right sides of the speakers and the wall should be the same. There should be no debris in front of the speakers. 2. The tweeter of the speaker should be at the same level as the listener's ear, and the angle between the listener and the two speakers should be 60 degrees, and there should be a certain amount of space behind the listener. 3. The walls on both sides of the two speakers should be consistent in acoustics, that is, the reflection of sound waves by the walls on both sides should be the same. 4. If the directionality of the sound waves of the speakers is not wide, the two speakers can be placed slightly inward. 5. For small speakers, if you feel that the low frequency is not enough, you can place the speakers close to the corners.
7. What should I pay attention to when connecting audio equipment?
The connection between the various levels of audio equipment is very important. If the connection is improper, it will not only affect the playback effect of the equipment, but may even damage the equipment.
A. Basic requirements for equipment connection:
(1) Signal level matching: When connecting audio equipment, you must pay attention to the difference in input and output signal levels between the equipment. If the level of the input signal of the pre-stage equipment is too large, nonlinear distortion will occur. Otherwise, the signal-to-noise ratio of the playback system will be reduced, and even the amplifier of the next stage equipment will not be driven. Therefore, when matching, you must pay attention to the level difference between the equipment. If the signal level is not matched in actual use, the input signal level must be reduced through the attenuation circuit, or the input signal level must be increased through the amplification circuit. For general dynamic microphones, the output voltage is a few millivolts, so a first-stage amplifier circuit is required to amplify the signal and send it to the preamplifier circuit. For record players, CD players and LD players, since their output signal levels are above 0.755~1V, they can be directly sent to the preamplifier.
(2) Impedance matching: In Hi-Fi audio equipment, for example, the output impedance of transistor power amplifiers is low impedance, while the output impedance of tube power amplifiers and other equipment is high impedance. If the impedance is not matched when they are connected to the speaker, the output power of the amplifier will be unevenly distributed, or the transient characteristics of the speaker will be deteriorated due to excessive damping. There are generally two types of impedance matching connections: balanced and unbalanced. The so-called balanced type means that the impedance of the two-core shielded wire transmitting the signal to the ground is equal. The so-called unbalanced type means that one of the two-core shielded wires is grounded. When the balanced output is connected to the unbalanced input, matching must be performed by adding a matching transformer.
B. Connector connection method: In Hi-Fi audio equipment, the connection of equipment is completed by various connectors. The commonly used connectors are as follows.
(1) Two-core plug: mainly used to transmit signals between various equipment and as an input plug for microphone input signals. According to its diameter, it is divided into three types: 2.5mm, 3.5mm, and 6.5mm.
(2) Lotus plug: Mainly used as input and output plug for lines between audio equipment and video equipment.
(3) XLR plug: mainly used for connecting microphones and amplifiers.
(4) Five-pin socket (DIN): Mainly used for connecting cassette recorders and amplifiers. It can concentrate stereo input and output signals on one socket.
(5) RCA plug: RCA plug is mainly used for transmitting video signals in equipment.
(6) F and M plugs: They are mainly used for the input and output of radio frequency signals in audio-visual equipment.
8. What is an "OFC" audiophile cable? What are "6N" and "7N" audiophile cables?
"OFC" is the abbreviation of "Oxygen Free Copper" in English, which means "oxygen-free copper". As we all know, gold and silver have the lowest resistivity and the best conductivity among metals, but if gold and silver are used as materials for making audiophile wires, the price is very expensive and is not acceptable to most audiophiles. Copper, as a commonly used metal material, has good conductivity and is widely used, but copper contains more impurities, most of which are oxides, which affects the conductivity of copper. At present, the most commonly used is the "OFC" wire, which is called "intelligent audiophile wire". It uses high-tech methods such as electrochemical method, PN junction implantation method, isotope irradiation modification method, etc. to change the metal structure of copper, so that the surface of the copper wire produces a unique metal structure, so that the surface of the same copper wire is suitable for transmitting frequency signals above 5000Hz, while its center is only suitable for transmitting frequency signals below 5000Hz, so that high and low frequencies do not interfere with each other, which is conducive to improving the clarity of the reproduced sound and the sound quality of the reproduced sound when transmitting large signals.
"6N" and "7N" are used by audiophiles to indicate the purity of audiophile cables made of oxygen-free copper. Because the English letter "9" begins with the letter "N", for the sake of convenience, audiophiles use "N" to represent "9", and the number before "N" indicates how many "9s" there are. For example, "99.9999%" can be represented by "6N", which means that its purity is 6 9s, and the number before N is 99.9999%.
The larger the number, the higher the purity of the audiophile cable.
2. What is the difference between a combination audio system and an audio system combination?
Answer: The so-called combination audio is what is usually called a package machine. The various equipment in its audio system has been selected and matched by the manufacturer and cannot be disassembled at will. In order to cater to the needs of most consumers, the manufacturer has given more consideration to the beautiful appearance and diverse functions of the audio system it produces, and the requirements for components and circuit structure are general, so the sound quality of the playback is also general, and it is only suitable for general consumers. For audio enthusiasts and some audio professionals, the sound quality of the combination audio cannot meet their requirements. They believe that the sound quality of the high-end combination audio can only be of medium and low-end level. Therefore, enthusiasts often freely select and combine according to their own hobbies and the characteristics of the playback sound of various equipment, so that the playback sound of the equipment has a certain personality. This is the audio combination. The audio combination mainly focuses on whether the sound quality of the equipment can express the connotation of the music and certain contents required by enthusiasts, while the appearance and function of the equipment are secondary. Of course, to carry out the audio combination, you must also have certain knowledge of music, electronics, acoustics, etc., in order to make the audio combination the best and most reasonable.
3. Home Theater
1. Overview of Home Theater
In recent years, a strong AV whirlwind has swept abroad, and for a time, a more enthusiastic and popular AV fever has quickly set off in China than HI-FI [AV: A (AUDIO) audio, V (VIDIO) video]. This refers to creating a perfect home theater center at home - home theater. Home theater is to realistically present the sound effects that can only be enjoyed in the theater in your home. This is the product of the highly perfect combination of digital technology and analog audio technology today. You can taste Xiangling and listen to wonderful music in the colorful programs such as CD, LD (video disc), VCD (small video disc), DVD (digital video disc), VCR (video), BS (satellite reception), etc., or you can sing karaoke to your heart's content, and you can also enjoy the video discs with Dolby directional logic surround theater effects, enjoy the cowboy style on the vast North American grasslands, enjoy the gorgeous style of the Alps winter ski resort, and appreciate the thrilling gunfight fighting scenes; feel how wonderful the future world in the mythical science fiction film is... All these are the supreme enjoyment brought to you by the home theater. How to configure an ideal HI-FI audio system or home theater?
2. Basic configuration of home theater
First of all, we pursue high-definition visual effects, so we must have a high-definition large-screen color TV, generally 25-34 inches. Imported brands such as Panasonic, Sony, Hitachi, Toshiba, and Philips are everyone's first choice. Of course, you can choose our domestic TVs. If conditions permit, we can choose better rear-projection large-screen color TVs (50 inches), plasma TVs, and even projectors to form a real home theater. When enjoying high-quality pictures, how can we not pursue high-fidelity sound effects? And for this premise, we need a set of efficient sound combinations. Since we want to have visual enjoyment, we choose LD, VCD, DVD and other audio and video sources. In fact, we can buy compatible machines, which is economical and practical. Audio is the focus of home theater. You must choose an AV amplifier with Dolby directional logic surround digital processing (detailed introduction later). There are six speakers in the home theater, namely the front left and right speakers, the rear surround speakers, which are used to create an immersive surround sound field, and a center speaker to strengthen the dialogue in the film. A subwoofer must also be configured to feel the overwhelming momentum. What is different from the sound combination in the home theater is that the home theater can create an immersive feeling, and this feeling is the surround processing effect of the home theater. Let's introduce the surround sound field below. Surround sound is to reproduce the direction of each sound source in the original signal during playback, so that the audience has a feeling of being surrounded by sounds from different directions. The current surround sound includes: Dolby Surround, Dolby Pro-Logic, THX, Dolby AC-3 and DTS.
(1) The playback format of Dolby Surround Sound is still stereo, but the left and right channel signals are decoded through the matrix to obtain a surround channel.
(2) Dolby Pro Logic Surround uses a 4-2-4 encoding system, and the four channels produced provide accurate positioning.
(3) THX The so-called THX (Tom Holman's experiment) system is a home theater system developed by the American Lucasfilm Company. It can produce the effect of a movie theater in a general listening environment. The format of the THX system is designed for independent six-channel wide-screen stereo theaters. Compared with other sound systems, the most obvious feature of the THX system is that the sound is more natural and clear, with a strong sense of three-dimensionality, the positioning of sound and image is very accurate, and it can produce a full range of dynamic range and frequency response, so that the audience can listen to the same playback effect at any position in the listening environment. The listening environment set by the THX system is that the front left and right speakers are full-band main channels, and the middle channel speaker is located behind the screen, which can produce sound sources in the three directions of left, middle, and right to achieve accurate sound and image positioning. The surround sound field is produced by the two speakers at the back, which can create an ideal diffuse surround sound effect. In order to enhance the shock of low frequencies, THX also adds a subwoofer to produce a grand scene of a theater. Compared with other audio systems, the biggest feature of the THX system is its unique control circuit, which is mainly composed of a re-equalization circuit, a decorrelation circuit, and a timbre matching circuit. The input dual-channel signal is first decoded by the Dolby decoder, and then the re-equalization circuit compensates for the imbalance of the sound in different listening environments, thereby eliminating various noises in the signal. The decorrelation circuit then divides the surround sound into two unrelated outputs, which drive the left and right surround speakers respectively to produce a diffuse surround effect. In order to produce a complete sound field, the timbre matching circuit can transmit the sound as it is, so that the playback sound from the front main channel to the back surround channel maintains the same timbre. The playback frequency response of the left, center, and right channels of the THX system reaches 20Hz~20kHz; the frequency response of the surround channel reaches 100Hz~7kHz. THX has high requirements on decoders and speakers. The performance of the speakers in the left, center and right directions must be consistent. There are only a few companies producing THX products internationally, so the price is relatively high. If the THX system is used for playback, the playback software must be encoded according to the THX standard, otherwise the THX effect will not be produced.
(4) AC-3 surround sound is a new generation of Dolby digital surround sound developed by Dolby in 1991. (AC stands for Audio Coding) This Dolby AC-3 surround sound has 6 completely independent channels, full-band left, right, center, left surround, right surround, plus a subwoofer channel below 120Hz, so it is also called 5.1 channel. In the AC-3 specification, the subwoofer is 10dB louder than other full-band channels to obtain shocking low-frequency information. AC-3 can also use the strong sound pressure of other channels to mask the noise of other channels. Due to this masking effect, Dolby AC-3 can achieve unprecedented digital audio compression efficiency, making the sound quality more realistic. The digital sound effect includes a wider dynamic range, all channels have a frequency response of more than 20KHz, a higher S/N ratio, and completely independent 6-channel high-power output, without the problem of weak rear surround output. Performance comparison between Dolby AC-3 and THX THX is a playback system that presents Dolly surround recorded sound with better effect (same as George Lucas' studio), which is basically Dolby surround's four-channel, that is to say, the rear surround sound is still a monophonic sound image with only 7kHz frequency response, not stereo. THX just uses unique equipment to do some processing: add subwoofer output; simulate surround sound into stereo; make corrections to the high-pitched area. Dolby AC-3 uses a new sound processing system from the beginning of recording -- 5.1 channels. Compared with the subwoofer output of THX and the subwoofer of AC-3, the subwoofer of AC-3 is an independent channel recorded with bass effect during the recording process, and its content is completely different from the main five channels; while the subwoofer of THX is separated from the original four channels by analysis, and it is not a special sound effect recorded on a separate track. There is a big difference between the two. AC-3 was launched in pursuit of more realistic sound effects that are more faithful to the director's intentions. It is a product of the new era and will not immediately replace the Dolby Surround decoder. There will inevitably be a buffer period for the two new and old systems to coexist, but future home theaters will definitely be based on Dolby AC-3.
(5) DTS (Digital Theater Systems). DTS uses a 5.1 multi-channel movie stereo system (five channels: front left, front right, center, left rear surround, right rear surround, plus a subwoofer channel), which is output by six speakers. It uses compressed digital signals with a compression ratio of 3:1. The 20-bit audio signals of the six channels are preset in the 16-bit PCM signal space of the two channels. When playing, they must be decompressed and restored to a 6-channel analog signal. Therefore, the smaller the compression ratio, the clearer and more detailed the restored signal will be, and it is closer to the sound heard in person. Therefore, the dynamic effect of the DTS sound field and the layering, continuity, width performance of the detailed sound and the 360-degree surround effect are superior to Dolby AC-3. Many music fans are still unfamiliar with the DTS system, but in fact, DTS has long been the new favorite of audio-visual players. DTS technology was founded by Mr. Deliber in January 1993. After this technology was developed and announced in the film industry, it was immediately highly valued by the great director Spielberg and Universal Pictures, who decided to first try to use DTS technology in the large-scale science fiction movie "Jurassic Park". Because DTS is also one of the audio standards of DVD, some DVD players have built-in DTS decoders when they are produced, which can directly play DTS discs and obtain the effect of 5.1 movie stereo. However, DTS is a unique audio encoding form, so only machines with DTS decoding function can play DTS-encoded discs. However, if your CD or DVD can use optical fiber or coaxial digital cable, you can use external connection to DTS decoder or pre-stage or amplifier with DTS decoding function to enjoy the unprecedented audio and video charm brought by DTS, rather than just ordinary two-channel effect.
(6) What is DSP sound field processing technology? "DSP" stands for "Digital Sound Field Processing". It is a new sound field processing system developed and produced by Yamaha Corporation of Japan in the 1980s. The so-called sound field processing technology is to process the sound wave reflection and reverberation signals of various singing and performance scenes to form different sound field characteristic data, which are packaged in a large-scale integrated circuit (DSP). When replaying, the DSP circuit is used to call out the corresponding simulated sound field data, so that various scene effects can be simulated more easily. Therefore, if the DSP scene signal is added to the signal that already has Dolby directional logic decoding, the sound field created will be more magnificent. DSP software is divided into two categories: those with scene characteristics and those with surround characteristics. The former is used to process the dialogue and background music of the characters, and the latter is used to produce surround sound effects. At present, DSP software generally has the functions of these two software. On AV amplifiers with DSP sound field processing, there is generally a "NATURAL SOUND DIGITAL SOUND FIELD PROCESSING" logo. At present, there are two main types of DSP sound field processing circuits. The most commonly used one is the "serial processing method", that is, the left, center, right and surround sound signals generated after processing by the Dolby decoder, only the surround sound signal enters the DSP processing system, and the surround sound processed by the DSP can produce various simulated sound field effects. The other is the "serial control method", that is, two DSP systems process the left, center, right sound field signals and surround sound signals respectively, so that the simulated front and rear sound fields are mutually expanded, thereby generating a complete simulated sound field. Its playback effect is better than the "serial processing method", but the circuit is more complicated. DSP sound field processing technology can generally produce the following simulated sound fields: Hall A in Europe: European Concert Hall A (2500 seats) Hall B in Europe: European Concert Hall B (2000 seats) Hall C in Europe: European Concert Hall C (1700 seats) Hall D in USA: American Concert Hall D (2600 seats) Hall E in Europe: European Concert Hall E (round 2200 seats) Live Concert: Live Concert Church: Church effect Large Chapel; Large Chapel Church effect Afterglow: Party effect Real Room: Standard listening room Space Flanger: Highlight space effect On the Town: Town street effect Rock Concert: Rock Concert Jazz Club: Jazz Club Concert Video 1: Concert Video 1 Classical/Opera: Classical/Opera Recital: Solo Recital Concert Video 2: Concert Video 2 Pop/Rock: Pop/Rock Music Pavilion: Medium-sized Stadium Tv Theater: TV Theater Mono Movie: Mono Movie Variety/Sports: Sports Program Movie Theater 1: Cinema 1 70mm Spectacle: 70mm thriller movie 70mm Musical: 70mm music movie Movie Theater 2: Cinema 2 70mm Adventure: 70mm action movie 70mm General: 70mm drama movie Dolby Pro Logic Surround: Dolby directional surround sound.
(7) What is the SRS system? How does it work? Generally speaking, only when the number of channels of playback reaches six or more can the human ear obtain the accurate spatial distribution of the sound source. Although Dolby surround sound has greatly improved the sound image positioning effect of the playback sound, its software must be Dolby encoded to obtain the Dolby effect, and the price of the Dolby playback system is also relatively high. The biggest feature of the SRS is that it can produce a sense of presence and surround stereo that is basically the same as the actual live effect from ordinary two-channel programs with only two speakers, and it is not limited by the listening environment. SRS has no requirements for the sound source. It can process single-channel, dual-channel, Dolby-encoded software and other sound sources and then replay them. The basic principle of SRS is to utilize the auricle effect of the human ear, that is, when replaying, no matter where the speaker system is, the human ear feels that the sound is coming from the spatial direction corresponding to the frequency response, regardless of the location of the speaker. Based on this principle, the SRS system uses circuits to modify the reproduced sound, compensating for the difference between the frequency response of the reproduced sound and the frequency response of the human ear, so that the reproduced sound forms a complete sound field reproduction system in terms of human psychology and subjective feeling.
3. AV amplifier
AV amplifier is the system center of home theater. It is an audio-visual device that integrates audio and video signal control. Compared with common amplifiers, AV amplifiers mainly have Dolby directional logic surround decoding, AC-3, DSP, digital sound field processing, AM/FM digital tuning and radio, as well as multiple audio and video IO interfaces. Some AV amplifiers also have SVIDO interfaces.
4. Speakers
The speaker is the most important part of the whole system. The performance index of the speaker (1) Sensitivity If the speaker is marked with 87 dB, it means that when a 1W pink noise is input to the speaker, the sound pressure value received at one meter in front of the speaker is 87dB. From another perspective, the sensitivity reflects the difficulty of driving the speaker. The sensitivity is best above 87dB. Such speakers are easier to drive and the requirements for the power amplifier are not too high. (2) Impedance refers to the resistance value of the speaker at a frequency of 1Kz. It is usually 4 or 8 ohms, of course, there are also 5, 6, 10 ohms, etc. In fact, the impedance of the speaker changes with the operating frequency. It is usually low in the low frequency band and high in the high frequency band. In the ideal state of the speaker, the smaller the change with the operating frequency, the better. (3) Power handling When we buy speakers, we usually see the words "how many watts to how many watts" on the nameplate. Its meaning is: the previous value refers to the minimum continuous power to drive the pair of speakers. Only when this power is reached can the speaker enter the best state and the index meet the requirements. The number behind it refers to the maximum power that the speaker can withstand. If the power exceeds this, the unit may burn out. (4) Frequency response refers to the frequency range in which the speaker can work. We generally require a full frequency band, that is, 20 Hz~20KHz, but there is usually attenuation at both ends. Of course, we require the frequency response to be as wide as possible, but it must also be flat. At least the attenuation at both ends does not exceed 3dB to be meaningful. (I) Left and right channel main speakers (20Hz~20KHz): During playback, it mainly reflects the size and depth of the front sound field of the audience, and expresses the left and right front sound field signals in the playback sound field. It plays a leading role in the playback process. When playing a feature film with Dolby decoding in the AV system, the left and right channel main speakers are to express its background music. It is required to be placed at the same height as the TV. (II) Center speaker (40~20k) The center speaker mainly expresses the dialogue of the characters and the sound in the middle during playback. The center speaker is usually placed on top of the TV. There are two types of center speakers: single bass and double bass. The former should be placed vertically, while the latter should be placed horizontally. (III) Surround speakers (40~10k) Mainly used to reproduce the sound behind the playback field. Only with surround speakers can the surround feeling of the sound field be reflected. Especially when playing war movies, when the plane flies from the back to the front, the performance of the surround speakers can make the audience feel immersive. The height of the surround speakers is usually 70 cm higher than the listener. (IV) Subwoofer (50~150) When reproducing dynamic signals or climaxes in feature films, the playback of the subwoofer can make the audience feel a kind of overwhelming momentum. Since the bass has no directionality, the subwoofer can be placed at will. Generally, the subwoofer is placed between the main channel and the surround speakers.
4. Introduction of world famous brand equipment
AV amplifier: DSP: MARANTZ SR73. SR82. SR92 KENWOOD KR-V6070. KR-V7070 KR-V8070 YAMAHA RXV-890 RXV-690 RXV-2090 PIONEER V504 TEAC 3020 JVC 508 808 DENON AVC-2000B AVC2500B AVC-2800 THX: KENWOOD KR-X1000 ONKYO TX-SV828 AC-3: PANSONIC KR-TX1000 YAMAHA DSP-3090 KENWOOD KR-V3090 DENON AVC-3800 AVR3600
Domestically produced: Arden, Bada, Tianyi, Zhongshen, Dongpeng, Desay, etc.
Equipment:
1. American speakers such as JLB, EV, BOSS, AR, ACOUTECHLABS, BOSTON, SOURCE, and equipment such as INFINITV, MCINTOSH, Lion, and Crown. Its characteristics are large dynamic range, good performance of music strength and dynamics, and more suitable for playing music with a strong sense of rhythm.
2. British speakers such as TANNOY, WHAREDWARO, B&W, MISSIO, NROGERS, MONITOR, and KEF, and equipment such as HARBETH, CELESTION, NAD, and QUAD. The characteristics of British equipment are: delicate expression of musical details, fine manufacturing technology, and more suitable for playing symphonies.
3. Japanese equipment in China is mainly packaged, such as Sony, Kenwood, Sansui, Aiwa, etc. The equipment mainly includes: ACCUPHASE, DENON, TEAC, YAMAHA, ONKYO, MARANTZ, AKAI, LUXMAN and CEC, etc. The characteristics of Japanese equipment are high manufacturing technology and more functions, especially in AV equipment, which has more advanced technology and is more suitable for playing pop music.
4. German ELAC, VISATON, HECO, etc. The playback sound of German audio equipment is relatively soft, the performance of music is relatively natural, and the production process of the equipment is relatively high. 5 Danish equipment mainly includes: AVANCE, DANTA, DYNAAUDIO, DALI, JAMO, etc. Its characteristics are that the manufacturing process is also relatively fine, and its playback can better express the connotation of music.
6. Canadian equipment has a relatively small share in China. Currently, the popular ones are PSB speakers, which have average craftsmanship but good performance and clean sound. Of course, the quality of various audio equipment is mainly determined by personal preferences and understanding of music, and cannot be generalized.
Speaker:
Tannoy in the UK VIFA in Denmark
French Proud Poetry
Domestic: Nanjing in Nanjing, Feile in Shanghai, Xinzi in Henan, Zhujiang in Guangzhou, Yindi in Shanghai, Huiwei in Zhuhai, Xinda in Chengdu, etc.
Speaker cable: Mecca Luhua (Japan), Orsonic (Japan), Budweiser (USA), ELEO (Germany), Sandfish (UK), Thunder (China), Teflon (China), etc.
Signal line: Monster (USA) AR (USA) Economical and practical Akihabara (Japan. Assembled in Shenzhen)
Material selection:
Capacitor: Capacitor is one of the most important components in the audio system. To obtain better sound quality, we must use good capacitors. For example, Japan ELAN high-speed electrolytic capacitors are the most commonly used capacitors and are also recognized as the best electrolytic capacitors. Panasonic gold electrolytic capacitors, Japan Chemical capacitors, Dutch Philips capacitors (good elasticity and most suitable for filter capacitors), Rubycon ruby capacitors (very common capacitors on the market), and West Germany's ROE capacitors. Polypropylene capacitors such as Germany's WIMA, Sweden's RIFA, the United States' EC, France's MPK (Thomson), etc.
Resistors: The resistors are the original high-precision five-ring resistors from Philips of the Netherlands. Potentiometers also play an important role in audio. We can choose Japanese ALPS potentiometers, which are recognized as fine products in the audio industry. Taiwan's Zhengxin 16 type and domestically produced Fengzhisheng are also good choices.
Transformer: We use a toroidal transformer, and if conditions permit, we can use an E-type transformer.
The signal line inside the machine must be shielded.
We use Japanese TDK for filters.
It is best to use gold-plated connectors to ensure good contact. The solder wire should be high in content, preferably silver.
Headphones: world-famous brands such as Deepsea (the cheapest one is over 300 yuan), Beyerdynamic from Germany, Triangle (Japan), Boss (USA). All are world-class professional headphones. Some Japanese headphones suitable for us include Sony, Panasonic, AIWA, etc.
5. Some information about audio DIY
Several high-performance amplifier chips suitable for our production:
TDA1521/TDA1514A
TDA1521/TDA1514A are two chips designed and launched by Philips of the Netherlands for low distortion and high stability of digital audio during playback. Therefore, the sound quality of the direct output of the CD player is particularly good. The parameters are: when the voltage of TDA1521 is ±16V and the impedance is 8Ω, the output power is 2×15W, and the distortion is only 0.5%. The working voltage of TDA1514A is ±9V~±30V. When the voltage is ±25V and RL=8Ω, the output power reaches 50 W and the total harmonic distortion is 0.08%. The input impedance is 20KΩ, the input sensitivity is 600mV, and the signal-to-noise ratio reaches 85dB. The circuit is equipped with waiting and quiet noise states, overheating protection, low offset voltage and high ripple suppression, and extremely low thermal resistance, with excellent high-frequency resolution and low-frequency strength. The sound is transparent and pure, the bass is full and thick, the treble is clear and bright, and it has the charm of a tube. The above two amplifiers have relatively few peripheral parts. They are "fool-proof" amplifier chips, which are very suitable for assembly by beginners. As long as you follow the circuit diagram, you can get a good effect without debugging. Because the input level of the chip is relatively low, we do not need a preamplifier in production, just connect it directly to our computer sound card, optical drive, and walkman. The famous computer multimedia speaker Edifier also uses these two chips.
LM3886
LM38863TF is a high-power audio amplifier chip launched by NS (National Semiconductor) in the early 1990s. The main parameters of this chip are: the working voltage is ±9V~±40V (recommended ±25V~±35V) and the continuous output power reaches 68W (peak 135 W) when RL=8Ω. If it is connected to BLT, the output power can reach 100W, and its distortion is less than 0.03%. Its internal design has a very complete over-consumption protection circuit. I am also using this chip. Its tone is very sweet, the sound quality is mellow, and it has the charm of electronic tubes. It is suitable for playing softer music. NS also has LM1875, LM1876, LM4766 and other chips that everyone is familiar with. Among them, LM4766 is the latest, with a dual-channel design, and contains overvoltage, undervoltage, overload, overtemperature and other protection circuits. Its output power is not less than 2×40W. The bass is deep and elastic, and it has the style of a tube amplifier.
TDA7294 is a rather innovative DMOS high-power integrated amplifier circuit introduced to the Chinese mainland by the famous European SGS-THOMSON STMicroelectronics Company in the 1990s. It swept away the raw, cold and hard sound of previous linear integrated amplifiers and thick film integration, and is widely used in the HI-FI field: such as home theaters, active speakers, etc. The design of this chip focuses on the sound quality, and has the advantages of bipolar signal processing circuits and power MOS. It has the characteristics of high voltage resistance, low noise, low distortion, and extremely friendly playback sound quality; short-circuit current and overheating protection functions make its performance more perfect. The main parameters of TDA7294: Vs (power supply voltage) = ±10~±40V; Io (peak output current) is 10 amperes; Po (RMS continuous output power) is 70W when Vs=±35V, 8Ω, 70W when Vs=±27V, 4Ω; music power (effective value) is 100W when Vs=±38V, 8Ω, 100W when Vs=±29V, 4Ω. The total harmonic distortion is extremely low, only 0.005%. In addition, SGS-THOMSON STMicroelectronics has several representative power amplifier chips, such as: TDA7295 TDA7296 TDA7264, TDA2030A (our commonly used Mai Lan subwoofer uses this chip), etc.
LM4610N
LM4610 is a high-quality DC-controlled audio circuit from National Semiconductor. It is a stereo integrated circuit that uses DC voltage to control tone, volume and channel balance, and has 3D sound field processing and equal loudness compensation functions. The circuit has smooth control, natural and smooth sound quality, clear high frequencies and good resolution. The 3D surround sound field it produces has a strong sense of three-dimensional space and envelopment, and the subjective feeling is similar to the effect of SRS. The main electrical parameters of LM4610N are as follows: It has 3D sound field processing function and loudness compensation function. Loudness compensation is aimed at the decrease in the sensitivity of the human ear to high and low frequency signals when the volume is low, so the high and low frequency ends are appropriately increased and compensated at different volumes, so that the human ear can always hear a flat and balanced response at any loudness. Its voltage range is: 9V~16V (typically 12V, current 35mA); distortion is only 0.03%; signal-to-noise ratio is up to 80dB; bandwidth is 250kHz, volume adjustment is 75dB; balance adjustment is 1~20dB; tone adjustment range is ±15dB; maximum gain is 2dB; LM4610N has the advantages of high input impedance (30Ω) and low output resistance (20Ω). Using LM6410N tone control circuit has a prominent effect on improving sound quality and strengthening low-frequency strength and three-dimensional space. It can be said that LM4610N is a fine product for assembling power amplifier system or replacing tuning part.
BBE Technology
BBE is a patented technology for sound enhancement and improvement. Its full name is Barcus-Berry Electronic, which is a new technology launched by the American BBE.sound company in 1985. It has been widely used since its appearance, such as Panasonic and Sony abroad, TCL, Skyworth, Lehua and other new generation color TVs in China. BBE technology is also used in recording and records, and some radio stations such as the Canadian Broadcasting Corporation, Swiss International Broadcasting, Korean Broadcasting and Japan's NHK government-run radio and television systems have applied this technology. High-resolution BBE circuit XR1075 XR1075 is the latest high-resolution BBE chip launched by the American XEAR company. It is based on XR1071 and adopts new bipolar technology to make the chip have lower noise coefficient and smaller total harmonic distortion. The chip is smaller, the peripheral components are further simplified, and the high and low frequency extension, high frequency resolution enhancement adjustment range and low frequency compensation range are wider than XR1071. High frequency adjustment range is -0.5~+13 db, low frequency compensation adjustment range is -0.5~+13db.
Digital Subwoofer Processor M51134P
M51134P is a special subwoofer detection and enhancement circuit developed by Mitsubishi Corporation of Japan for AV audio-visual systems. It includes: frequency detection, adjuster, level detection, low-pass filter VCA voltage-controlled amplifier, etc. The principle is to use digital filtering to detect the level of the low-frequency components in the input signal, strengthen the corresponding low-frequency components and perform low-frequency dynamic expansion (completed by the voltage-controlled amplifier). Its principle is different from the general low-pass filter form of the subwoofer enhancement circuit. The subwoofer effect provided by M51134P has a strong sense of shock, especially thunder, cannon, explosion, etc. M51134P only detects signals below 120Hz. If there is no component below 120Hz in the input signal, there is no output.
The latest standard virtual Dolby surround sound chip QS7779/QS7785
QS7779/QS7785 is a single-chip virtual surround sound processing circuit launched by Canada's Qsound Audio Laboratory. It is currently recognized by the industry as the virtual Dolby surround chip with the processing effect closest to natural original sound! QS7779 is a 2-input 2-output method, and QS7785 is a 2-input 5-output method. Both include Dolby Pro Logic and DVD (AC-3) mixed signal decoders. They use Qsound Laboratory's patented Qsurround virtual surround technology and are authorized by Qsound Laboratory. The main functions of this chip are: (1) If the input is a normal stereo signal, the stereo effect is enhanced; (2) If the input is a 2-channel matrix encoded signal (Dolby Pro Logic or mixed AC-3 signal), it is first decoded and then virtualized into a 2-channel or 5-channel output. QS7779 main features: 1. With Dolby Pro Logic and DVD (AC-3) mixed signal decoder output, virtual surround sound is achieved using 2 speakers. 2. Signal-to-noise ratio 11db, dynamic range 110db. QS7785 main features: 1. Built-in Dolby directional logic and DVD (AC-3) mixed signal decoder output, the decoded surround signal is 2-channel full-band, the same as AC-3 surround sound, better than Dolby directional logic system. 2. The front uses 3D stereo enhancement technology, and the rear uses 3D synthetic virtual surround technology, which is divided into two enhancement modes (low enhancement and high enhancement), with center output and bass enhancement functions. 3. Use 5 channels to achieve surround sound, and 2-channel output can also be used. 4. Signal-to-noise ratio 11db, dynamic range 110db
Op amps (operational amplifiers) We often see or use the following: 4558 (cheaper and generally used in some walkmans). NE5532 was once known as the king of operational amplifiers. AD712K.AD827 (very good op amps are hard to buy in the market. I heard that it takes three months to place an order. The market price is about 100 yuan per piece). The above are all dual op amps, and there are also quad op amps such as TL084.LT058, etc.
What is Dolby Technology
From a technical perspective, Dolby's technologies can be divided into analog and digital technologies. From a usage perspective, they can be divided into professional and civilian technologies - sometimes the same technology is often used in both professional and civilian fields. The following paragraphs briefly introduce these technologies. Technologies: A-type SR (spectral recording) Dolby Stereo AC-1 Dolby E B-type S-type Dolby Surround AC-2 C-type HX-Pro Dolby Pro Logic Dolby Digital (AC-3)
A-type (A-type noise reduction)
Dolby's first technology was Dolby A-type noise reduction technology, introduced in 1965. It was designed for professional recording studios to record low-noise masters. In the early to mid-1970s, its application expanded to movie recording studios and movie release copies to obtain better movie sound effects. Almost every studio uses Dolby Laboratories' A-type noise reduction equipment when making analog masters, whether for movies or other purposes.
B-type (B-type noise reduction)
Dolby B-type noise reduction technology is a simplified version of A-type noise reduction introduced in 1968, which can reduce noise by 10dB at high frequencies. This technology expands the application of Dolby Laboratories' technology to the civilian industry, enabling consumer electronics manufacturers to produce cassette tapes and recorders with low noise performance for consumers. Dolby produces professional B-type noise reduction encoding equipment that can be provided to tape duplicators to produce encoded civilian B-type noise reduction tapes. In addition, this technology is also licensed to consumer product hardware manufacturers, who can choose to purchase Dolby B-type noise reduction integrated circuits (ICs) from different semiconductor manufacturers for the production of audio decks, portable single players, car recorders, etc., all of which can record and/or play back cassettes with Dolby B noise reduction encoding.
C-type (C-type noise reduction)
Dolby C-type noise reduction technology was introduced in 1981 and is the second generation of civilian systems developed by Dolby - the basic feature is that its noise reduction ability is twice that of B-type noise reduction, while adding other technical features (such as spectrum shift, anti-saturation, etc.). This technology has been licensed to dozens of electronics companies and is used in almost all home cassette recorders and advanced portable players. Like B-type noise reduction technology, Dolby also produces C-type noise reduction professional encoding equipment for tape duplicators.
SR (spectral recording)
Dolby SR technology was introduced in 1986 as the second generation of professional recording systems. It is designed not only to provide higher noise reduction capabilities, but also to provide many of the latest technical components to expand the dynamic range of recordings, making the recorded master indistinguishable from the actual sound. Therefore, SR is considered a signal processing technology rather than just a noise reduction technology. Dolby produces SR equipment for the recording and film industries.
S-type (S-type noise reduction)
Dolby S-type noise reduction technology is derived from Dolby SR. Like SR, it has the same functions as fixed and non-fixed frequency bands, anti-saturation, spectrum offset and modulation control. As a licensed technology of Dolby Laboratories (although Dolby produces professional S-type noise reduction encoding equipment), it can ensure 24dB high-frequency noise reduction and 10dB low-frequency noise reduction. Advanced cassette recorders basically have Dolby S-type noise reduction technology, which can make consumers' self-recorded cassettes have the effect of CDs.
HX Pro
Dolby HX Pro technology was introduced in the early 1980s to provide high-frequency extension peak reserve to improve recording quality, using a dynamic adjustment method for recording bias level. This technology is used in professional and consumer equipment (all licensed production).
Dolby Stereo
After introducing A-type noise reduction to the film industry, Dolby's next major contribution was Dolby Stereo. This technology allowed filmmakers to use matrix technology to have four channels of sound information (left, right, center, and surround) on the release print, and for theaters to restore the four channels of information when showing it to the public. Dolby produced equipment to record Dolby Stereo films, and every film can use these equipment and have the corresponding Dolby technology support. Dolby also produces theater playback equipment, which has been sold all over the world.
Dolby Surround
Dolby Surround is the application of Dolby Stereo technology in home products. Audio and video companies are authorized to record the same four-channel matrix encoding information on VHS tapes and discs as on movie copies. Consumer electronics manufacturers are authorized to produce civilian surround sound decoding equipment that can reproduce four channels in the home.
Dolby Pro Logic
Dolby Pro Logic is the second generation of Dolby licensed home surround systems. One of the biggest advantages of Dolby Pro Logic is the use of an effective center channel and accompanying speakers. Old stereo systems produce a virtual center channel that requires the listener to sit directly in front of the TV screen. If the listener's seat is off center, the dialogue will sound off center. But with Dolby Pro Logic and a properly placed center speaker, the dialogue will always be positioned toward the screen, and the left and right stereo speakers can be placed more widely for expanded music and other sound effects. The Dolby Pro Logic decoder can also better decode the surround information, which is presented by a pair of speakers placed slightly behind the listener on the left and right sides.
AC-1
AC-1 is Dolby's first digital audio coding technology. Some system vendors first adopted this technology in 1984 when the theory of reducing bit rate was just proposed. AC-1 is a more accurate form of adaptive delta modulation (ADM), in which the change of signal amplitude over time can be transmitted instead of the absolute value. In addition to encoding the change of amplitude, a system of pre-emphasis and de-emphasis of dynamic range is also applied to reduce the audible coding noise. At that time, the digital signal processing methods we commonly use today did not exist. AC-1 was envisioned for commercial television broadcasting. The solution was driven by a relatively complex encoder and used a simple and therefore cheaper decoder. This technology is used in satellite and cable transmission systems. Dolby Laboratories has its own codecs manufactured and sold, and also licenses this technology to other manufacturers to produce and sell codecs.
AC-2
Dolby AC-2 technology is an adaptive transform coding algorithm based on perceptual coding, which achieves high-quality audio at a lower bit rate, thus greatly reducing the data capacity in satellite and terrestrial lines and digital audio storage media. This digital algorithm developed by Dolby Laboratories adopts a multi-band approach based on the psychoacoustic masking effect. The bit allocation method is 80% fixed configuration and 20% adaptive configuration, which makes the complexity of encoding and decoding relatively low. Dolby produces professional codecs (such as Dolby FAX) that apply AC-2 technology, and also licenses this technology to other manufacturers for use in their products.
Dolby Digital (Dolby Digital AC-3)
Dolby Digital (AC-3) is an advanced sensory coding technology used to transmit and store up to five full-band channels of information, plus a low-frequency effects channel (often referred to as .1 channel because of its small amount of information and the number of bits required), and the total space required is even less than the space of one channel of PCM linear coding on a CD. Compared with AC-2, Dolby Digital is a more powerful and flexible coding system that can provide a range of functions:
1) Mixing function enables compatible playback of mono, stereo, directional logic and full 5.1 channels;
2) Transmit information such as dynamic range control and dialogue level control to the decoder;
3) It has multiple bit rate operation modes. Dolby Digital technology can be enjoyed in thousands of movies and the latest generation of LDs. Dolby Digital technology is also used in DVD audio tracks. It is the audio standard of the American New High Definition Television System (ATSC) established in 1998, and became an optional audio format of the European Digital Television Standard (DVB) in 1999.
Dolby E
Dolby E audio coding enables an AES/EBU audio pair or a pair of soundtracks on a digital video recorder (VTR) to carry up to eight channels of broadcast-quality audio information for post-production and distribution. Unlike Dolby Digital, which is the encoding method for delivering audio to the end listener in the home, Dolby E allows programs to be edited and decoded and re-encoded multiple times without audible sound loss. Dolby E also carries important audio metadata, or data about the audio, throughout the distribution editing process. Dolby E equipment became available in the summer of 1999.
History of Dolby Laboratories
Dolby Laboratories was founded by Dr. Ray Milton Dolby. Dr. Dolby was born in Portland, Oregon, USA in 1933 and grew up in the San Francisco Bay Area. When he was 16 years old and still in high school, he worked for Ampex, one of the earliest manufacturers of tape recording equipment in the United States. Later, he was responsible for developing the electronic circuit part of the world's first practical video recorder developed by the company.
In 1957, Dolby graduated from Stanford University and received a Marshall Scholarship from Cambridge University in the UK to study long-wave X-rays. In 1961, he received a doctorate in physics. In 1963, he accepted an appointment from the United Nations to go to India as a consultant for two years.
As an amateur recording enthusiast, Dr. Dolby had long recognized the deterioration of recording quality caused by background noise when recording audio or video signals on tape. While in India, he began to seriously consider a way to reduce noise without compromising recording quality. His explorations became the basis for the future Dolby A-Noise Reduction, B-Noise Reduction, and C-Noise Reduction systems.
After returning to England in 1965, he set up his own laboratory in London to implement the ideas he had come up with in India. In 1968, the company named "Dolby Laboratories" was established. Although the center of work was in the UK for the first 10 years of the company's establishment, it has always been an American company. In 1976, the company's main work moved to San Francisco.
In 1965, a Dolby A-type noise reducer (A for Audio) was produced. The system was designed to address a variety of audio noise reduction applications, especially the noise generated by tape recorders when recording masters in recording studios. By 1966, several noise reduction technologies had been introduced, but they all compromised the quality of recordings to some extent. Therefore, the difficulty Dr. Dolby faced at the time was how to convince industry insiders and potential customers of his technology. At that time, multi-track recorders, from 4-track, 8-track, 16-track to 24-track, began to be used. When multi-track recorded tapes were mixed, the noise level of the mixed two-track master was much higher than the master recorded directly on the two-track.
In January 1966, the British department of Decca Records believed that the Dolby A-type noise reducer could indeed work as described by Dr. Dolby, so they ordered 9 Dolby A301 A-type noise reducers, which were first used to record some Mozart piano concertos played by Ashkenazy in Vienna in May 1966. In November 1966, Decca published the first record recorded with the Dolby A-type noise reducer - Mahler's Second Symphony conducted by Solti.
Subsequently, the recording industry began to recognize and use the Dolby A-type noise reduction system extensively. At first, it was only used to record classical music, but when multi-track recording technology became popular, it was widely used. Soon, professionals and non-professionals around the world began to associate "Dolby" with high-quality recording.
There was a growing demand for Dolby to invent noise reduction technology for consumer tape recorders. At the urging of KLH, an American commercial tape recorder manufacturer, Dolby Laboratories began developing a more practical civilian noise reduction technology in April 1967, which was initially called the "Simplified Dolby System" and later became the widely known Dolby B-type noise reduction technology. When the development of Dolby B noise reduction technology was nearly completed, Dr. Dolby decided that Dolby Laboratories would not produce civilian audio products or consumer electronics products, but would license Dolby's technology to manufacturers for application and production by already mature manufacturers. By the end of 1974, Dolby Laboratories had 47 authorized manufacturers, including all major manufacturers of consumer audio equipment.
Since then, Dolby Laboratories has developed a series of technologies: C-type noise reduction, SR (spectral recording), S-type noise reduction, HX Pro, Dolby Stereo, Dolby Surround, Dolby Pro Logic, AC-1, AC-2, Dolby Digital (AC-3), Dolby E. These technologies are widely used in professional and civilian audio equipment, film recording, theater playback equipment, digital broadcasting and other aspects.
In addition to its headquarters in San Francisco, Dolby Laboratories currently has branches or liaison offices around the world: Los Angeles, Wootton Bassett (UK), London, Brisbane, New York, Tokyo, Shanghai, and Beijing.
What is HDCD?
(Excerpt from "High Fidelity Audio"/Yin Haibin)
HDCD is the abbreviation of High Definition Compatible Digital. It uses a new recording technology. When the analog audio signal on the master tape is sent to the HDCD encoder, it is encoded into a digital signal with a high resolution exceeding the traditional CD format of 44.1KHz and 16bit. The signal generated at this time will be more than what an ordinary CD can accommodate.
Highly compatible with high-resolution HDCD
CD Status
The 12cm CD laser disc has been around for more than a decade. It is still the main sound source for Hi Fi equipment due to its many unique advantages such as small size, easy storage, wide frequency response, high signal-to-noise ratio, and large dynamic range. As people's appreciation of music improves, the inherent defects of CD sound sources are becoming more and more prominent. Compared with traditional LP records, the sound played by CDs always has a sense of stiffness, less details, and lack of presence. If the sound quality of VCDs, which has become popular in recent years, is also included, many audiophiles and experts will continue to regret it.
As for the inherent defects of CD, we have to start with the Red Book specifications that were formulated for CD back then.
Due to the limitations of the microprocessor technology software and hardware at the time, the CD DA laser disc red book standard released in February 1982 made the following provisions: disc diameter 120mm, disc speed 1.2m/s, modulation mode EFM, error correction CIRC, data rate 0.6Mbps, data volume 0.7GB. If you want to record the changing analog audio signal on this CD, you must first sample the analog signal. The condition for reproducing the signal waveform is based on Shannon's theorem: let the signal bandwidth be Bw and the sampling frequency be fs. If the condition Bw<=fs/2 is met, the original waveform can be fully reproduced. Based on the research result that the highest frequency audible to the human ear is 20kHz, the sampling frequency of the CD is 44.1kHz. The sampled values are discretely digitized relative to the amplitude (i.e. quantized) to obtain a series of pulse trains. After adding CIRC error correction code, synchronization signal and address information, the data information obtained after EFM format modulation can be recorded on the CD record.
Due to the limitations of the capacity of the laser disc and chip technology at the time, quantization adopted 16-bit operation, and the dynamic range it could express was D=20lg2＋1.76[dB]=98dB(n=16), which is the theoretical dynamic range of the CD.
With a frequency response of 20kHz, a dynamic range of 97dB and unmeasurable jitter, laser turntables have become a shining star in the field of digital audio. In a short period of time, they have become an important source of sound for Hi-Fi sound equipment, so that people have abandoned tapes and vinyl records without hesitation. However, with the further development and exploration of digital audio, the defects of this 44.1kHz/16bit recording format have become increasingly prominent.
First of all, the 44.1kHz sampling rate is the first factor that affects the sound quality and timbre. The 44.1kHz sampling can completely reproduce a 20kHz sine wave, but it is difficult to completely reproduce a 7kHz non-sinusoidal signal. This is because the non-sinusoidal signal can be decomposed into a fundamental wave plus the second, third, and third harmonics. Although the fundamental wave can be reproduced, the third and higher harmonics may be lost or distorted after D/A conversion, so that the final waveform is different from the original information, resulting in a change in timbre.
Due to the knowledge and conditions at that time, the data information recording format of the laser disc was defined as 16 bits, and the theoretical dynamic range that could be achieved was 98dB. In fact, a safety margin was left to avoid strong limiting, and the 16 bits could not be fully utilized. In addition, the loss in the recording encoding and decoding process made it difficult to break through 96dB, which was obviously not enough for expressing classical percussion (118dB). This is a kind of distortion unique to digital audio that people have discovered - subtractive distortion.
Since the original analog information is infinitely changing, the information on the laser disc is recorded in 65536 stages. In order to complete the information, the 16-bit CD recording has to round up the sound between the processing stages and add it to the previous stage or the next stage. In this way, even if the information contained in the CD can be completely restored, there will be errors compared with the original sound.
If the quantization accuracy is high, the original analog information can be reproduced more realistically and with richer details. It is easy to draw a conclusion by comparing the screens of a 16-bit game console and a 32-bit game console. Low-bit quantization makes the error after quantization larger than high-bit quantization. These errors (quantization noise) after quantization make the sound harsh and muddy, especially for small signals. These harmonic components that are not in the original signal constitute additive distortion.
As a special case of digital audio, the sound quality of VCD is a typical result of digital calculation. Compared with ordinary CDs, it feels more hollow, lacks details and layers, and has more prominent high-pitched piercing. This is because VCD can play back image and sound information on a 12cm disc. It uses the masking effect of the human ear to ignore the information that is not easy for people to perceive, and performs a large amount of compression and encoding reorganization on the data. The process is a large-scale subtraction operation, and the final result is similar in form but lacks in spirit.
If high-bit and high-sampling rate digital processing is used, the sound quality can achieve a qualitative leap. In fact, many recording companies have used 96kHz sampling rate, 20-24bit recording technology to produce master tapes in the early stage of CD production. However, when making CD records, they are restricted by the current CD specifications and have to re-encode to conform to the 16bit/44.1kHz format. Therefore, the CD turntables marked with 20 or 24bit that we can see are actually still 16bit data streams.
If we want to change the current situation of CD, we must first overturn the existing CD format and adopt high sampling, high bit recording format and playback equipment, which will undoubtedly increase the information capacity and transmission speed. The current CD player is not up to the task, but fortunately the advent of DVD has solved this problem. However, the recording format of high-quality audio CDs has not yet been determined, and once it is determined, it means that CD turntables, DACs, LDs, and VCD players that have been popular in the market for more than a decade will no longer be able to play with it and will become toys. Even CD players worth tens of thousands of yuan will not be able to escape the fate.
Another way to solve the problem is to improve the previous CD to achieve a breakthrough under the current system, just like the transition from black and white television to color television. HDCD technology is a successful and mature example of this kind of solution.
HDCD Overview
In order to improve the defects of the existing CD recording format and make it highly compatible and have a breakthrough in sound quality, Pacific Microsonics of the United States launched the patented HDCD recording and broadcasting technology, whose full English name is High Definition Compatible Digital, which is translated as high-resolution CD. Laser discs encoded and manufactured using the HDCD method are highly compatible with ordinary CDs. When played on ordinary laser players, the superiority of HDCD encoding and recording technology can be appreciated. If played on a CD player with HDCD decoding function, the unique charm of all released HDCD information can be fully appreciated: clear and delicate sound quality, wide dynamic range, extremely high signal-to-noise ratio, and more natural and realistic sound.
Encoding and Production of HDCD
In response to the limitations and shortcomings of the traditional CD recording format, the two founders of HDCD at PM, recording engineer Keith O. Johnson and computer expert Michael W. Pflaumer, have found and confirmed several key factors that affect CD sound quality during many years of audio production, and proposed practical solutions.
HDCD technology emphasizes the integrity and accuracy of the recorded signal in the early stage of recording production. It uses a sampling frequency of 88.1kHz, which is twice the normal sampling frequency, to sample the analog signal in order to maximize the high-frequency response and reduce lossy distortion. The high sampling rate also leaves enough room for HDCD encoding operations.
The sampling value is 1677216 when quantized with 24 bits, which is 256 times higher than that of the 16-bit system. The use of high-bit processing technology can improve processing accuracy, reduce quantization errors, and increase the dynamic range to 120dB.
In the process of analog to digital signal conversion, HDCD technology attaches great importance to conversion accuracy, minimizes crosstalk and processing stability, and can achieve conversion accuracy of one millionth and distortion component <-120dBfs.
This high-precision, wide-band digital signal forms the basis of HDCD encoding and production, and its data information volume is very large. It cannot be accommodated by the conventional CD PCM encoding format. If it is to be compatible with ordinary CD players, it must be specially encoded.
The use of high sampling and high bit technology to record CDs has been generally recognized and widely adopted, but one thing to remind you is that the 20-bit and 24-bit CD laser discs currently available on the market are actually the number of bits used in the recording process. Due to the 44.1kHz/16bit standard format established by the CD "Red Book", these high-information master tapes are recalculated and encoded into 16-bit CD records when recording CD records. Therefore, the specifications that our CD players can interpret are still 16bit/44.1kHz. Due to the different methods used by various record companies in the conversion process, the sound quality of different versions of CDs that we can hear now is indeed different, but one thing is certain: the sound quality of CDs produced by high-bit high sampling technology is far better than CDs produced by 16bit/44.1kHz recording format.
So how does HDCD technology produce high-definition records that are compatible with ordinary CDs?
Sampling frequency conversion. First, the 88.1kHz sampling data is dynamically converted, which is a major feature of HDCD technology. It uses multiple data interpolation filters to be dynamically controlled by the analysis system. This system analyzes the signal bandwidth, peak energy and high-frequency information in real time, and accurately controls the wave pass characteristics of the filter with high-resolution signals. The execution result is that even if the final sampling rate is changed to 44.1kHz, its bandwidth still changes very little between 16kHz and 22kHz. The system has a record of exceeding the 44.1kHz sampling rate and can reflect every subtle change in the sound.
Amplitude analysis. Another feature of HDCD technology is the effective control of amplitude. The signal transmitted by the Decimation filter is a 24-bit/44.1kHz signal. In order to accommodate this signal, the encoder is accurately amplitude analyzed and gain controlled at this level, quantized and edited to 20 bits and then distributed to 16-bit format for operation.
The range of sound changes in nature is very wide. The sudden sound pressure can cause the recording equipment to be overloaded and clipped. In the process of analog tape recording, level compression is used to avoid tape saturation distortion. For a digital recording system, overload can lead to unnecessary quantization errors (data fragments), which will also affect the sound quality. For this reason, ordinary A/D converter equipment has an absolute maximum recording level (0dB) to ensure that the peak is not clipped. HDCD uses a unique amplitude encoding technology to obtain one bit more capacity (equivalent to +6dB) than conventional digital recording to process large dynamic signals. Due to the use of digital operation processing, this extended information can control the decoder recovery of the playback device with precise and stable characteristics. Coupled with the unique "look ahead" capability of digital processing, the system can instantly restore the gain before a large signal begins, providing greater information capacity to avoid instantaneous signal overload.
For this one-bit information expansion, the operation time is subject to the implicit control code of HDCD (discussed later). For ordinary CD playback, the information remains unchanged. However, when playing with an HDCD decoder, the information can be accurately expanded under the control of the implicit code, thereby achieving the purpose of large dynamic playback.
High-frequency dithering technology (Dither). The use of high-frequency dithering technology can improve the resolution of quantized signals, improve the nonlinear transformation characteristics of the quantizer, reduce the harmonic distortion of low-level signals, and may reproduce signals below the quantization difference. However, if added improperly, high-frequency oscillation (dither) will become real added noise. HDCD technology uses improved high-frequency dithering technology, making the music details richer and the noise inaudible.
HDCD implicit control code. For the final quantization operation part of HDCD, in order to accurately control the excessive information recorded by HDCD encoding to be played accurately on the decoder, a relevant control code is set. This code is inserted into the least significant bit LSB in the word group segment of the data record. If it is played by an ordinary CD player, the code is implicit and not activated. Due to its specific position and only occupying 1% to 5% of the LSB bit, the impact on the CD sound quality is weak and inaudible. When playing with an HDCD decoder, the system can accurately capture the implicit code and use it to activate the information of the main data channel, so that the amount of information expands and obtains information output several times that of the ordinary CD format. After DA conversion, an analog audio signal with large dynamics, rich details and high signal-to-noise ratio can be obtained.
To avoid erroneous operation, HDCD uses a dual code synchronization timer set in the main and sub channels, so that it is accompanied by the main information in the word group segment and the timing will not be misplaced. The main channel selection data is only valid when the hidden code echoes the main related code, otherwise the decoding operation is canceled.
After the analog audio signal is low-pass filtered by the buffer, it is first converted to digital, and a high-frequency disturbance signal is used to control the ADC in real time, quantizing it to generate an 88.1kHz, 24-bit data stream, which flows to the main and auxiliary channels. The main channel information is delayed and stored, while the auxiliary channel information is analyzed one component ahead of the main channel to generate a control signal. The signal dynamically controls the digital filter to perform sampling rate conversion, amplitude coding and gain control. Finally, the microprocessor separates the information that is easily lost during analysis, filtering and data reformatting (this information may involve timbre, sound field, and subtle sounds), and combines it with the control code to generate a hidden code that is inserted into the LSB of the main channel audio data. After high-frequency disturbance processing, it is quantized to 16bit/44.1kHz standard CD format output, completing the full HDCD encoding process.
HDCD decoding process and PMD100
The decoding operation of HDCD is the reverse action of the encoding process. The design purpose is to replace the digital filter of DAC with HDCD decoding dedicated integrated circuit to complete the dual functions of HDCD information decoding and oversampling digital filtering.
The decoder first detects whether the LSB bit in the data stream carries the HDCD hidden code. If so, it activates the main channel audio data information to expand it according to the continuous instructions of the hidden code, and restores the compression of the data information during the encoding process. Due to the control of the hidden code, the peak can be accurately expanded in time, and the information below the average level value can be appropriately reduced in gain. Therefore, the HDCD method can obtain high definition of large dynamics and small signals higher than the conventional one.
The only HDCD decoding chip is PMD100 produced by PMI of the United States. This chip needs to be used with authorization. It is a large-scale integrated circuit in a 28-pin DIP package.
When PMD100 receives input data in HDCD encoding mode, it automatically switches to HDCD decoding format and outputs current at its 27th pin to drive the LED light-emitting tube as a status indicator.
When it is not an HDCD signal, the information data is received and processed by conventional oversampling digital filtering, so the device has dual characteristics. When used as a normal CD format digital filter, the device also has excellent characteristics, with a passband ripple of no more than 0.0001dB from 0 to 20kHz and a stopband attenuation of >120dB.
Other features of the device are:
·With 2, 4, 8 times oversampling digital filtering
Can accept 24-bit input data and process with the same precision
· Accepts any input sampling frequency from 32 to 55kHz
Output 16, 18, 20 and 24 bit different data formats
With digital de-emphasis function
Digital volume control with 0.188dB steps
· Clock frequency is 256fs or 384fs optional
·Two squelch modes: soft and hard
Provides two control modes: hardware setting and program mode.
Provides 8 different types of high-frequency dithering modes to suit different types of DACs
Provides a constant output clock to the DAC, ensuring that the DAC output is free of offset and pulse generation even if both the input data and the main pulse are lost
The pin arrangement of the decoding chip PMD 100 is similar to that of some top digital filters, such as SM5842, SM5803, DF1700, etc. Therefore, on a DAC or CD player with the above filters, an ordinary CD player or DAC can be converted into a processor with HDCD decoding function by making slight modifications.
SACD (Super Audio CD)
1 SACD Introduction
1. What is SACD?
The advent of the CD can be considered one of the most exciting events in the field of music reproduction, bringing musical performance to a new level. In recent years, some demanding listeners have put forward higher requirements for the sound quality of the reproduced sound. For this reason, Philips and Sony (the inventor of the CD) have developed the Super Audio CD (SACD). SACD uses the most advanced DSD technology to make the playback sound quality even better, and restore all the details of the live performance vividly. This technology is close to reproducing the original analog waveform, resulting in clear and more natural sound, showing the presence and emotional communication of the musical performance extremely realistically. Using a sampling frequency 64 times higher than that of CD, SACD can reproduce extremely pure and natural sound - whether it is two-channel stereo or multi-channel recording.
2. Why develop SACD?
SACD is a logical evolution of CD, and it brings a bright future to the audio industry. SACD composite discs bring many development opportunities to every CD-related industry from music albums to retailers. This is because SACD composite discs can play sound with excellent sound quality on SACD players, and this disc can also be played normally on any existing CD player.
2. Comparison of SACD functions with other related audio formats
See Table 1.
┏━━━━━┯━━━━━━━━━━┯━━━━━━━━━━━━━━┯━━━━━━━ ━━━━━━┓
┃To all new voices│ │ │ ┃
┃Audio formats │ SACD │ DVD-Audio │ DVD-Video ┃
┃ Ask│ │ │ ┃
┠─────┼───────────┼───────────────┼──────────────┨
┃Application fields│Home, car and portable│Home, car and portable and future│Home, car and portable and future┃
┃ │ and future PC drivers │ PC drivers │ PC drivers ┃
┠─────┼───────────┼───────────────┼──────────────┨
┃ │Independent multi-channel signal with better quality than CD│Independent multi-channel signal with better quality than CD;│Independent multi-channel signal with worse quality than CD┃
┃ Performance│Channel and separate stereo mix│Stereo signal in most cases comes from│Stereo signal in most cases comes from┃
┃ │Signal│Multi-channel mixed signal│From multi-channel mixed signal┃
┠─────┼───────────┼───────────────┼──────────────┨
┃ │Completely compatible with ordinary CD players│Not compatible with ordinary CD players, compatible│ ┃
┃ Compatibility │ Compatible with DVD-Video players │ DVD-Video players (when AC-3 content is available │ Not compatible with ordinary CD players ┃
┃ │Player (CD content) │Time) │ ┃
┠─────┼───────────┼───────────────┼──────────────┨
┃Anti-piracy measures│Strong series of copies│A series of copy protection mechanisms│A series of copy protection mechanisms┃
┃ and copy protection │ Protection mechanism │ │ ┃
┠─────┼───────────┼───────────────┼──────────────┨
┃Value-added content│Lyrics, pictures and videos│Lyrics, pictures and videos│There is a wide range of value-added content potential┃
┠─────┼───────────┼───────────────┼──────────────┨
┃│Navigation-style search like CD│On the additional TV screen, through the menu│On the additional TV screen, through the menu┃
┃Navigational search│Search. Enhanced│structure of additional content for navigational search of CD-ROM.│structure for navigational search of CD-ROM. ┃
┃ │CD-style navigation search. │ │ ┃
┗━━━━━┷━━━━━━━━━━┷━━━━━━━━━━━━━━┷━━━━━━━ ━━━━━━┛
3. SACD Format Introduction
Philips and Sony are the most innovative pioneers in sound recording and playback technology, both in the civilian and professional fields. For example, in the early 1980s, they successfully launched the CD, the world's first civilian digital audio format. Based on pulse code modulation (PCM) technology, CD integrated the most advanced technologies at the time and became the most user-friendly civilian format for playing music. Soon, CD became the most successful recording medium for pre-recorded music programs. So far, 800 million CD players have been sold worldwide, and software sales have exceeded 14 billion discs. Despite such impressive achievements, people have been exploring digital audio technology with higher resolution. The goal is to find a digital music medium that can meet the very high quality requirements in the music field and satisfy the most demanding listeners.
To achieve such a goal, a completely new audio recording and playback solution is required. For this purpose, Philips and Sony proposed a solution such as SACD. This solution uses direct stream digital (DSD) signal processing technology. This signal processing technology was originally developed for digital archiving of precious analog master tapes.
DSD is a processing technology developed based on the 1-bit F-signal modulation principle. It uses a sampling frequency of 2.8224MHz (equivalent to 64 times the 44.1kHz sampling frequency used by CDs). This technology enables SACD to have a usable bandwidth of 100kHz and a signal-to-noise ratio of 120dB. The final DSD data stream is closer to the original analog signal than the PCM modulated data stream.
The DSD modulation solution eliminates several processing steps in the signal processing chain (see Figure 1).
Figure 1 Comparison of PCM and DSD solutions
SACD uses DSD technology on all channels. In a multi-channel disc, there are a total of 8 tracks (6 channels for multi-channel and 2 stereo channels for dual-channel) stored on the disc. The technical specifications of SACD stipulate that the audio signals in all channels are completely discrete. It is precisely because of the use of DSD technology as the technical basis of SACD that Philips and Sony can provide the civilian market and the music industry with sound with unparalleled sound quality.
One of the key factors to achieve SACD sound quality is its multi-channel capability. Some producers intend to faithfully restore the acoustic characteristics of the performance site by adding channels. Others intend to develop new creative mechanisms to enable the audience to experience a 360° sound field. This can make the audience feel more emotionally touched and give musicians more room for artistic creation. Multi-channel SACD is replacing two-channel stereo SACD and is becoming another way for people to enjoy music.
Figure 2. Multi-channel audio system setup
But without the development of optical storage media technology, it would be impossible to achieve this level of sound quality. By using shorter wavelength lasers and special optical devices, the capacity of optical discs has increased seven times compared to the 1980s. In fact, SACD uses DVD disc technology as the physical data carrier.
4. Composite discs and compatibility
1. Data structure of composite optical disc
The SACD format can identify composite discs. This composite disc cleverly combines the usual "Red Book" format CD content onto a single 4.7GB recording layer. This concept is the basic connection between the SACD format and the mature CD format (Figure 3). The composite disc contains a standard-precision CD layer and a high-precision SACD layer. The SACD layer can contain up to 4.7GB of DSD data. The logical structure of the SACD layer is relatively simple and as clear as the CD content layer. The use of the value-added data area in the SACD layer is optional. This area can contain content such as lyrics, video images and pictures. Figure 4 shows the content of the composite disc.
Figure 3: Composite disk is fully backward compatible
2. Physical structure of composite optical disc
From the outside, the SACD composite disc is no different from any other disc with a diameter of 12cm and a thickness of 1.2mm. However, from the inside, this disc is made of two 0.6mm data recording layers glued together, one of which contains the SACD data and the other contains the CD data. The CD data layer is close to the disc label side, while the SACD data layer is located in the middle of the disc. When picking up the CD data, the SACD layer is actually undetectable by the reading laser beam. The data contained in the CD recording layer of the SACD composite disc is fully compatible with the "Red Book" CD standard, so this composite disc can be played on all machines with CD playback function.
Figure 4 Contents of the composite disc
3. Bonding of composite optical discs
Using polycarbonate as the bonding material, the bonding process of the composite optical disc is the same as that of DVD5. However, if cycloolefin is used as the bonding material, in order to obtain the appropriate bonding force, the surface pretreatment of the photosensitive substrate is required. This pretreatment process can be completed on the production line.
4. Testing equipment and technical requirements
During the production of hybrid discs, online monitoring systems with similar standards as for DVDs can be used, and offline parameter detection is also required. For the measurement of HF and servo signals of CD and SACD layers, commercial SACD test equipment produced by most test equipment manufacturers can be used. Usually, these signals are measured to the same standards as DVD (SACD layer) and CD (CD layer of hybrid disc).
5. Production of composite optical discs
The production of composite discs is similar to that of DVDs, but there are some differences. Any existing DVD production line can be converted to produce SACDs. All plastic discs absorb moisture in humid environments, and this absorption is only slightly more severe for composite discs than for CDs and DVDs. Label text with a protective coating can better prevent moisture intrusion than the laser-see-through side. Necessary protection measures must be taken to prevent composite discs from curling in humid environments. There are two common solutions: one is to use a material that does not easily absorb moisture for the recording layer; the other is to add a front coating.
6. Brief introduction to the manufacturing process of SACD composite disc
(1) Master and template: The production of SACD composite discs requires two sets of pressing templates, namely CD and SACD templates. Both templates are formed from the master disc of a 413nm (DVD) laser beam recorder. In the process of manufacturing the glass master, some special measures for copyright and anti-piracy are added. After that, the processing of the master and template is the same as the production process of DVD.
(2) Photosensitive substrate: The CD and SACD templates are used to replicate the photosensitive substrate of the composite disc. For both recording layers, replication is done using a conventional 0.6 mm photosensitive substrate (DVD) template and sputtering equipment.
Easily create your own DTS CD
---- Author: Shen Sheng
Whenever you enjoy the climaxing music in a DVD, have you ever thought about collecting these beloved music clips and enjoying them separately? Or when a friend has a top-notch DVD audio test disc and refuses to lend it, you want to take it back to test your own audio equipment? For some music lovers, they often encounter the same problem. For example, ever since I saw the DVD of "Fantasy 2000", I have always wanted to extract the exciting music from it and make a CD to enjoy it slowly with my eyes closed; not only that, I often want to burn all the thrilling multi-channel special effects in "The Prince of Egypt", "Terminator 2" and "The Matrix" on a disc so that I can take it to an audio and video equipment store to test the machine...
All of this is now easily achieved. All you need is a DVD-ROM and an ordinary CD burner, and then follow the methods in this article to burn your favorite multi-channel audio tracks from the DVD to a CD-R (RW), and make a DTS CD to take home for listening and testing.
DTS and DTS CDl
First, let's take a look at what DTS CD is:
The so-called DTS soundtrack uses the 5.1 channel standard after being encoded by DTS (Digital Theater Systems). The maximum encoding flow of DTS is the same as that of LPCM, that is, 48000*16*2=1536000bps=1536kbps per second. Relatively speaking, this is much larger than the 448kbps provided by Dolby Digital (AC3), another popular multi-channel encoding system. In other words, DTS has much less distortion in the compression process of multi-channel sound products. This feature gives DTS an inherent advantage in multi-channel encoding systems. Because of this, DTS-related products and equipment are increasingly favored by consumers in the market.
As for DTS CD, as the name implies, it is an audio-CD with DTS soundtrack. Its storage method is the same as that of ordinary music CDs, which are 16 bits and 44.1kHz sampling frequency. However, the actual content recorded in DTS CD is not the PCM sampling signal of ordinary CD, but the DTS encoded soundtrack signal. Since the DTS CD file recording format is compatible with ordinary CDs, it can be reproduced, produced and played in the same ordinary way as music CDs. In addition, most DVD players and CD players with digital output functions on the market currently support direct output of DTS digital signals. In addition, most audio enthusiasts or equipment stores already have DTS decoding equipment, which makes it easy to enjoy DTS CDs.
Extract multi-channel audio tracks from DVD files
To make a DTS CD from a 5.1 channel audio track on a DVD, you need to first copy the digital audio signal from the DVD to the computer hard disk. This is because the DTS encoding and CD burning work must be completed on the computer, and copying the digital information to the hard disk is the most basic condition for completing the DTS CD production. Therefore, we also need to have a basic understanding of the DVD data structure.
If we use a DVD-ROM to open a DVD video disc, we will see a file directory named "VIDEO_TS". The main video, audio, subtitle data, time code, paragraph segmentation information, etc. in the DVD video disc are all stored in various VOB files under this "VIDEO_TS" directory. Naturally, the multi-channel AC3 audio tracks in the DVD video disc are contained in these VOB files.
It is not complicated to separate the various audio tracks contained in VOB. Many software can complete this task. Here we take VOBRator as an example to roughly understand the basic principle of separating audio tracks. After you have a certain understanding of this step, you can apply it by analogy and use various advanced DeCSS software you like.
After opening the VOB file on a DVD disc (or hard disk) with VOBRator, a tree-like list of DVD track information will appear in the "Streams" column. This list lists all the contents contained in the VOB. When selecting each item, the information column on the right will also display the specific information of the track: such as sampling rate, number of channels, bit rate, etc. What we need is to let VOBRator output specific audio tracks according to our needs.
In the entire tree list, we must first select ****.VOB at the "root" and click the "include in output" option on the right to cancel it. Then you should see a cute little red cross on all the items. Then, select the 5.1-channel AC3 audio track you want to capture, and also click the "include in output" option on the right to make this track the only one selected. Now, VOBRator will obediently process the selected audio track according to your requirements. Click "DeMUX" (i.e. directly separate the data), and VOBRator will output a ****.ac3 AC3 format multi-channel audio track file.
You can use the same method to continue to capture your favorite audio track segments, and finally collect these AC3 files together, further encode them into DTS, and import them into CD burning software to further burn them into DTS CD.
Channel separation
After obtaining the 5.1-channel AC3 digital audio file, the next goal is to separate each channel in the digital audio file for re-encoding. There are a series of professional or amateur software that can be used to edit and separate the channels in AC3 audio files. Among them, the most famous one is the powerful Dobly Digtal AC3 editing software launched by Sonic Foundry, a professional audio editing software development company - Sonic Foundry Soft Encode.
Sonic Foundry Soft Encode can decode and encode AC3 audio channels, and we need to use it to decode AC3 files and output WAV audio files recorded in PCM format that can be recognized by the encoding software. Of course, PCM-WAV itself does not support 5.1 multi-channel mode, and DTS encoding software cannot recognize 5.1-channel AC3 files. Therefore, we must separate each channel in the AC3 file into a WAV file. Although this process is cumbersome, it is not difficult:
Open Sonic Foundry Soft Encode, first enter [Decode Setting] under the [Option] menu, and confirm that the "Dynamic Range Compression" is in the "RF remod mode" mode to ensure the correct decoding effect. Then click [Open...] in the [File] menu. Note that you must select "Decode to PCM" in the open mode conversion item next to the file type and confirm to open the AC3 audio file. This process may take a while, as the CPU needs to decode the AC3 file and save it in PCM format to a temporary folder. The waiting time is related to the CPU's computing speed. When the decoding is completed, the 6-channel audio tracks will be neatly arranged in the main window of Sonic Foundry Soft Encode.
At this time, if we are lazy and browse the directory where Sonic Foundry Soft Encode stores temporary files, we will see that there are 6 *.tmp files lying there quietly. They are actually six-channel audio files arranged in the order of "left", "center", "right", "left rear", "right rear", and "subwoofer". We just need to copy these files and rename them as WAV files, and then we can directly give them to the encoding software for use.
Of course, the lazy method above has a flaw, which is that it is easy to confuse the order of the six channels. Therefore, a proper method should be prepared as a backup: After the AC3 file has been opened, right-click on the channels that do not need to be retained one by one, and select "Delete" to delete them, so that Sonic Foundry Soft Encode only retains one "left front" channel. Click the "Save as..." icon to save this single "left front" channel as a standard PCM-WAV file for future use. After that, use the same method to save the "center", "right front", "left rear", "right rear", "subwoofer" and other channels separately.
Regardless of which of the above methods we use, after we obtain the separated channels, we can start encoding these 6 channels into DTS format.
DTS Encoding
DTS encoding software is rare, and the only software available to ordinary users may be Minnetonka's SurCode series. In the SurCode series, there is a "SurCode CD Pro DTS" that is specially designed for making DTS-CDs. Users only need to provide the WAV files of each channel to the software, and SurCode CD Pro DTS can easily combine and encode them into 44kHz DTS format WAV audio files that can be used for DTS-CD burning.
The interface of SurCode CD Pro DTS is also very simple and intuitive. We just need to fill in the output file name at "Destination...", and then add the PCM audio track and channel files just separated from AC3 one by one according to the prompts on the interface, and then we can immediately click the "Encode" button to start DTS encoding. SurCode CD Pro DTS will also directly output the "disguised" DTS-WAV format, making it more convenient for other CD burning software to burn DTS CDs.
l Burning
At this point, the key issue of converting multi-channel information into a format that can be used to burn DTS-CD has been solved, and the remaining work can be said to be very easy. The method and process of burning DTS-CD are exactly the same as burning ordinary music CDs. For example, using the popular burning software Nero, just create an Audio CD project, drag the collected DTS-WAV files into the Audio column on the left according to your preferences, and click the "Burn" button in the toolbar, and our first DTS CD is done. If it is a direct-drag burning software such as EZCD or Drop CD, it is even simpler. Just drag the DTS-WAV generated by SurCode CD Pro DTS into the CD icon.
With the possibility of copying and making DTS CDs by yourself, you can enjoy the thrilling feeling of 5.1 channels with fewer restrictions. As for the enjoyment of the soul, the cultivation of art, and even the fever of the equipment, you can do whatever you want without any constraints.
Notes
When listening to DTS CD, you need to connect the digital output (SPDIF or optical output interface) of the player to the corresponding DTS decoder or decoding amplifier, and then the corresponding DTS decoder or decoding amplifier will decode the digital signal to obtain the wonderful DTS 5.1 channel output. If you play on a computer, you need to install a DTS-supported player such as PowerDVD to correctly play back the multi-channel audio signal of the DTS CD.
Basic audio articles
DTS is a hot topic in the home theater market this year. AV receivers with built-in Dolby Digital/DTS dual decoding functions have become the new favorites in the market, and DVD players with built-in DTS 5.1 decoding output have also been launched one after another. Next, let's learn about the basic situation of DTS and its technical background.
DTS technology is closely related to the American gold medal director Spielberg and the American Universal Studios. The DTS headquarters is located in Hollywood, USA, and has branches in Brussels, Belgium and other places. The company's main structure is divided into two parts: the professional "digital theater system" mainly for the encoding and decoding of movie music recording sites and cinemas, and the consumer electronics "DTS technology" mainly for the development of home decoders and the introduction of DVD/LD/CD and other package software.
DTS technology was created by Mr. Dreberg in January 1993. In the early 1990s, when the digital soundtrack of movies was first proposed, Dreberg proposed the solution of using CD-ROM to record compressed sound signals. At that time, although the technology of writing digital sound signals directly to film had been developed, the reading method of this technology often resulted in reading errors and the stability was not ideal. However, CD-ROM has been widely used in the computer field, with a stable supply and low price. When using CD-ROM to record, you only need to install the CD-ROM with recorded audio digital signals when showing the movie, and there is no need to record on traditional film.
After the development of this technology, once it was announced in the film industry, it was immediately highly valued by the great director Spielberg and Universal Pictures, who decided to first try to use DTS technology in the large-scale science fiction movie "Jurassic Park". As a result, "Jurassic Park" conquered the audience with its realistic special effects and stunning digital surround sound effects, and set off a "Jurassic whirlwind" with a global box office revenue of 920 million US dollars, becoming one of the most popular movies in history, second only to "Titanic" in 1997 and "Star Wars Prequel" in 1999. Spielberg spoke highly of DTS technology, and Universal Pictures was also very optimistic about the prospects of DTS. Therefore, Spielberg and Universal Pictures even cooperated with Derryberg to establish DTS.
Since Spielberg first used DTS in "Jurassic Park", many Hollywood movies have also begun to use DTS's digital surround sound system. Many directors, production staff and mixers believe that DTS is the system with the best sound reproduction, and it can even compete with the original excellent 70mm film magnetic track soundtrack. Theater operators also believe that the movies produced by DTS are full of charm, and DTS technology is easy to operate, so it has gained absolute support. In the fifth year of DTS's advent, there are more than 200 movies with DTS soundtracks, and almost all new Hollywood movies have both Dolby Digital and DTS soundtracks. As the terminal for playing movies, the number of cinemas using DTS sound reproduction systems is increasing rapidly. Currently, more than 60% of cinemas in North America are equipped with DTS sound reproduction systems, more than 25% in Europe, and 45% in Asia. These numbers are still increasing. It can be said that DTS has become a leader in digital surround sound tracks alongside Dolby Digital and has received strong support from most cinemas in the world. Together with Sony's SDDS, it has become one of the three major digital surround sound systems in the film industry.
In the field of consumer electronics, the chip for DTS decoder was developed in early 1996. About a year later, many audio product manufacturers began to adopt this technology. By the end of 1998, there were 12 manufacturers authorized to manufacture independent DTS decoders or AV amplifiers with built-in DTS decoding functions. At present, almost all AV amplifiers on the market with a price of more than 4,000 yuan have built-in DTS decoding functions.
As for DTS software, in January 1997, DTS-LD discs of "Jurassic Park" began to be sold, especially those of UNIVERSAL, and later, DVDs of DTS movies also began to be sold. However, in terms of quantity, DTS discs still lag far behind Dolby Digital versions.
For LD, the data transmission rate of Dolby Digital is 448kbps, while DTS's signal compression ratio is more than three times the information volume of Dolby Digital, which is roughly the same as 16bit/44kHz/2-channel linear PCM, about 1411kbps, so DTS data occupies more space. In Japan, EFM digital sound is stipulated as the standard format for LD, and there is no space to accommodate DTS audio data. Therefore, Japanese versions of LD do not even have DTS versions. For DVD, the typical data transmission rate of 5.1-channel Dolby Digital format is 384kbps, and the transmission rate of DTS sound recorded on DVD is actually roughly the same as 48kHz/16bit/2ch linear PCM sound, reaching 1536Kbps, which is higher than the transmission rate of DTS signals recorded on LD and CD (1411kbps). The sound of DTS-DVD is generally sold in the combination of DTS 5.1 channels + Dolby Digital 2 channels. DTS is different from Dolby Digital in that it has 5.1 channels, while Dolby Digital can flexibly choose from 1 channel to 5.1 channels. Since DTS has a higher data rate than Dolby Digital, DVD movies in DTS format of about 2 hours often omit some functions such as multi-language and multi-angle or additional content such as film highlights and making-of specials. It is also difficult to have Dolby Digital 5.1 or PCM sound for selection at the same time, unless a larger capacity double-layer structure DVD disc or double-sided DVD is used to ensure the quality of the picture. Otherwise, it will not meet the requirements of directors and filmmakers for DVD quality. Therefore, except for "Saving Private Ryan", many works of the great director Spielberg (such as Jurassic Park, Lost World, etc.) have never met AV enthusiasts until recently, when the long-awaited "Jaws" special edition DVD was released.
The sound designers who participated in the DTS production process indicated that DTS can further expand its application in movies, music and other fields. Yael Hud of Enterprise studio believes that 5.1 channels are most effective for live sound sources and can create a sense of presence that cannot be achieved by 2-channel mixing of broadcast recordings. He has won the production award 7 times. As a Universal Studios that has always supported DTS technology, the sound design studio composed of Max Stroh and Clark Landy has 20 years of rich experience in film mixing and has produced famous blockbusters such as "Speed", "Waterworld" and "Twister". They said that the competitiveness of DTS and Dolby Stereo's LT, RT Master, Dolby Digital and SDDS first started with the mixing operation. The 6-channel MAC Master they designed is all formatted.
When all movies and theaters are replayed with digital soundtracks, bold mixing that enhances the surround sound effect is possible. The revolutionary significance of digital soundtracks in terms of sound quality, especially in expanding dynamics and ensuring high signal-to-noise ratio, is equally important. This is paid great attention to when making DTS Master. When encoding, the signals below 140Hz of the surround channel must be separated from the subwoofer channel. Of course, if this is not the case, the surround channel can also be set to the full range. The sound processing in "Speed" and "Twister" is almost carried out according to this setting.
Understanding and purchasing multimedia speakers
With the development of multimedia audio-visual, speakers are an important part of audio equipment and their importance is gradually being recognized by everyone. However, think about whether you have budgeted 700 yuan for the sound card and only 100 or 200 yuan for the speakers when you build or upgrade your computer. This is probably a common thing for most people. However, in the scientific configuration of multimedia equipment, what price range of speakers meets your needs, how to understand and choose speakers, and identify counterfeit products, this is exactly the problem that this article will solve for you.
The structure of the speaker. Ordinary multimedia speakers are mainly composed of the following parts:
1. Shell. Common speakers are mainly made of wood or plastic (some professional speakers are cast and filled with cement, steel or sand). Wooden speakers are made of composite medium-high density boards, and the thickness should be more than 10mm. Compared with plastic speakers, they have better anti-resonance performance, and the speakers can withstand greater power and are not limited in size. Plastic speakers are relatively cheap and are one-time molded products. They can be rich in design but are limited in size, relatively small, and can withstand relatively small speaker power, which is suitable for multimedia speakers. Poor-quality speakers are mainly caused by insufficient density of the density board, thin board, or brittle plastic with sand holes, which is easy to crack.
2. Power supply. The circuit inside the speaker is a low-voltage circuit, so first you need a transformer that converts high voltage into low voltage, then use two or four diodes to convert AC into DC, and finally use capacitors of different sizes to filter the voltage to make the output voltage smooth (in ordinary speakers, voltage regulator tubes are generally unnecessary). To be honest, these parts are not noticed by people, but their importance to the speaker is no less than that of the host power supply to your computer. The transformer is generally fixed at the bottom of the main speaker (this is also the reason why the main speaker is heavy), and it is required to have sufficient power output. Inferior products often cut corners here. Taking into account some loss and efficiency factors, it can be calculated that if the rated power of the transformer is 100W, the power of the amplifier chip that it can actually drive smoothly must be below 45-40W, so by calculating the power relationship between the speaker transformer and the amplifier, it can also verify whether the actual rated power of the speaker can reach the nominal value. The rectifier and filter capacitors are both on the circuit board. The large filter capacitor (several thousand microfarads) should be an electrolytic capacitor, and the larger the better. A large capacitor or two medium-capacity capacitors in parallel can be used to achieve filtering, while small capacitors (less than a few tenths of a microfarad) are used to make up for the lack of high-frequency filtering by large filter capacitors. The rectifier part is generally not a big problem, but in order to reduce costs, the filter capacitors of inferior speakers are obviously insufficient, and some are even less than 2000 microfarads; and small capacitors are likely to be ignored by them.
3. Power amplifier part. This part consists of the pre-stage operational amplifier and the post-stage power amplifier. The operational amplifier only plays the role of voltage amplification. In order to prepare for power amplification, it amplifies the voltage amplitude of the input signal to above the minimum value required for power amplification. In addition to the frequency range and distortion, the most important requirement for it is that the amplification factor must be sufficient. For the power amplifier chip, it can be called the core of the speaker. The key lies in its rated power. According to the standard, the rated power of the marked speaker should not exceed the typical value of the power amplifier chip. If there is a higher part, it is the "thriftiness" of the inferior speaker.
4. Circuits for special sound effects and functions. This part is not available in all speakers. A speaker may have one or several of them, or none of them. This includes the digital-to-analog conversion circuit of the USB speaker, three-dimensional sound field processing chips such as SRS, APX, Spatializer 3D, etc., active electromechanical servo technology circuits and BBE high-definition plateau sound playback system technology circuits, etc. From a technical level, small workshops cannot make USB speakers, so manufacturers who can make USB speakers have a certain level of technology and should be trustworthy. From the perspective of three-dimensional sound field processing technology, all the chips used are ready-made chips from abroad, and it is just a matter of circuit interface and position. What I want to say here is that some low-end speakers also have 3D sound field functions. Their 3D effects are not achieved by chips, but by an extremely simple feedback circuit. The effect can be imagined.
4. Speaker unit. Generally, wooden speakers and better plastic speakers use two-way frequency division technology, that is, two speakers, high and mid-range, are used to achieve sound playback in the entire frequency range; while some plastic speakers used as surround speakers on X.1 (X=2, 4 or 5) use full-band speakers, that is, one speaker is used to achieve sound playback in the entire sound range. Common mid-range speakers are mostly coated paper cones, ceramic paper cones, paper-based wool cones, loose-pressed cones, bulletproof cloth cones and PP cones. They have their own characteristics: paper cones have the advantages of simple and natural sound, low cost, easy to mix with other materials, and a combination of rigidity and flexibility. The disadvantages are poor moisture resistance and difficult to control consistency; bulletproof cloth cones have the advantages of wide frequency response and can withstand large power, which is the first choice for those who love strong bass, but because of its heavy material, it has poor playback effect on small and weak signals, and the effect of light music is not very good and the cost is high; wool woven cones are softer in texture and lighter in material. The vibration of the voice coil with the same energy will make its amplitude larger, so it has better performance for small and weak signals and perfect performance for soft music. The disadvantage is that the bass effect is not good, and the performance of rock music and marches is not satisfactory; PP (polypropylene) membrane has good toughness and consistency, low distortion, and good performance in all aspects, but the price is slightly higher; loose-pressed cones are mainly used in subwoofers, and the sound playback performance in the low-frequency band can be said to be a choice. The tweeters have silk membrane and PV membrane soft dome, which are used in mid-range and low-end speakers respectively. Since the speakers used in the multimedia field must be magnetically resistant, the design of the speakers adopts a double magnetic circuit, and a magnetic shield is added behind the speakers to prevent the leakage of magnetic lines. In order to save costs, the speakers of counterfeit speakers are likely not magnetically resistant!
5. About the design. Ordinary speakers can be divided into two types: bass reflex and sealed. A sealed speaker is a closed box with no openings inside that connect it to the outside world. A bass reflex has a cylindrical bass reflex tube installed on the front or back panel of the speaker to connect the inside and outside of the box. It works according to the principle of the Helmholtz resonator, and its advantages are high sensitivity, high power tolerance, and wide dynamic range. Since the sound waves on the back of the speaker can also be released from the bass reflex port, its sound efficiency is higher than that of a sealed box. The same speaker installed in a suitable bass reflex box will have a low-frequency sound pressure 3dB higher than that in a sealed box of the same volume, which means that it helps the performance of the low-frequency part of the speaker, so this makes the bass reflex box widely popular.
6. About subwoofers. For subwoofers, in order to ensure the performance of the speaker, the box must be strictly sealed, so a three-cavity design should be adopted, that is, there are two cavities behind the speaker. As shown in the picture, a foreign X.1 subwoofer adopts a two-cavity design of a single speaker, that is, there is only one cavity behind the speaker, plus a cavity in front of the speaker, so there are two cavities.
7. About flat-panel speakers. Several manufacturers, represented by Yasuda, have launched ultra-thin flat-panel speakers. It should be said that the advantage of this product lies in its versatility rather than the performance of the speaker. Because it is a flat panel, it is not as large as a traditional speaker, so its placement and installation are extremely flexible, and you can even stick it on the wall. However, in terms of performance, this product has no advantages at all, and its sound quality is even inferior to that of low-end wooden speakers, which is hard to praise. There is nothing magical about flat-panel speakers. Since it is a speaker, it must rely on the vibration of some substance to drive the vibration of air molecules and spread the sound. Here, this substance is a flat panel. What replaces the vibration of the flat panel? It is also a voice coil. In the final analysis, its principle is the same as that of ordinary speakers. It is achieved by the vibration of the voice coil to drive the vibration of the sound medium.
As a famous computer peripherals expert, Logitech's products such as mouse, keyboard, camera and even game controller are sought after by many computer users. In fact, multimedia speakers are also Logitech's "forte". However, although the previously launched Xuanyin and Shengmei series products are very distinctive in technology and design, they do not seem to have a great impact in the market, which also makes computer users seem a little unfamiliar with Logitech speakers.
In 2001, Logitech spent 120 million US dollars to acquire Labtec, a famous audio brand in North America. At the same time, it also acquired Labtec's 20 years of unique audio patent technology. After nearly two years of research and absorption, Logitech finally released a revolutionary audio product in 2003 that combined the technologies of the two companies - the Z series.
The Z series uses many exclusive patented technologies such as the continuous vented band-pass bass design (Series Vented Band-Pass), and is also equipped with the most popular advanced technologies in the audio industry.
Logitech Z-680 speakers are positioned as high-end 5.1-channel speakers and are THX certified. Z-680 speakers come with data cables for DVD players, CDs, PS2, XBOX and PC 2, 4, 6-channel sound cards. Z-680 speakers have built-in Dolby Digital and DTS decoders and support 5.1 channels. As a rare 6-piece sound system that has passed THX certification, Z-680 is not only the flagship of the series, but also the leader of this type of system currently on the market.
In fact, the Z-680 is very special from the packaging, because it seems that no other multimedia speakers have such a large size and weight. It takes a lot of effort to take it out of the box. In fact, the Z-680 has exceeded the application scope of multimedia and is more positioned as an entry-level home theater home audio system. Its various functions are also aimed at multi-functional applications in the home.
The Z-680 consists of eight components, including a subwoofer, four satellite speakers, a center speaker, a music control center, and a remote control. The overall appearance is silver-gray, which looks very modern and is very suitable for modern home environments.
When it comes to Swans speakers, I believe everyone will first think of its classic M200. But that was more than two years ago. After two years of hard waiting, audiophiles finally got Swans' new near-field monitor-level speaker - T200a. The T in the product number represents the speaker series with a simple black and white non-resonant triangle (Tri) geometric structure, and a represents the active system.
The two main boxes of T200a are exactly the same. The shape breaks through tradition and is extremely avant-garde. The main colors are black and white, with sharp contrast and great visual impact (see the title picture). The appearance breaks through the traditional cubic shape. The front panel is a polyhedron composed of three triangles, which has the flavor of fashionable electrical appliances, so it is easier to open a breakthrough in the market of families and individual users. The front panels of the entire speaker on three sides are black piano lacquer. A white panel is embedded in the middle of the front panel, and there is a round hole for an LED indicator below. When the speaker is working, a blue light is faintly emitted under the black mask, which not only decorates the speaker but also serves as a power indicator.
Speaker unit
The midrange unit of T200a is arranged at the top, while the tweeter is at the bottom. The midrange unit of T200a adopts the 5-inch M5N magnesium alloy long-stroke magnetic shielding speaker of Swans (see Figure 1), and the tweeter is the 25mm natural fiber braided membrane neodymium iron boron magnet tweeter of Swans.
The rated power of the M5N mid-bass unit is 35W, and the maximum power is 70W. Its diaphragm adopts an advanced aluminum-magnesium alloy integrated metal structure diaphragm, and the surface of the diaphragm is sprayed with a special damping layer, and the high-compliance support system has very good rigidity and dynamic stability. The M5N adopts a high-power, high-temperature resistant, eddy-current loss-free Kapton voice coil skeleton and a high-temperature resistant SV wire voice coil, a shielded magnetic circuit system and a long-stroke linear displacement design. The unique symmetrical magnetic field (SMD) driving technology of HiVi puts the voice coil in a symmetrical driving magnetic field, thereby obtaining a symmetrical and balanced driving force, which can significantly reduce the inductance of the voice coil to reduce the confrontation of the back electromotive force on the output stage. The control characteristics of the speaker are thus improved, the sound is fuller and more powerful, and the distortion is also reduced.
The TM1A-T tweeter uses a top-grade German natural fiber spherical diaphragm, with a flat frequency response, high frequencies up to 20kHz, and a sweet and delicate tone. The unit uses a shielded high-performance NdFeB magnet, and the sandwich anti-magnetic structure can effectively suppress the leakage magnetic field and create a miniaturized linear uniform magnetic field. TM1A-T uses an American liquid magnetic cooling high-temperature resistant aluminum alloy skeleton and copper-clad aluminum wire voice coil, with a large power bearing capacity and a strong sound-absorbing back cavity design, which reduces the nonlinear distortion of the unit near the resonant frequency.
Functional design and interface
In terms of functional design, besides the switch, T200a only provides the only adjustment of the total volume, but this is also a good thing for bookshelf monitor speakers. Because for a monitor-level speaker, any modification of the sound is redundant. T200a is quite professional in audio input. It has both balanced input and unbalanced input (see Figure 3). Our ordinary lotus head, that is, two-core RCA connector, belongs to the unbalanced input method, which is also the most commonly used connection method on our PC.
Design structure
T200a abandoned the design concept of main and sub-boxes in traditional multimedia speakers and adopted the design of a single-channel independent power amplifier. Therefore, the left and right boxes are exactly the same in appearance, box structure and internal circuit design. This can ensure that the sound characteristics of the two boxes are completely consistent, which is an absolute improvement. The T200a box is a phase-inverted design, with the phase-inverted port opening backwards and located below the heat sink on the rear plate. The box uses an inverted triangle that is slightly tilted backwards as the front sound baffle. The baffle has a small radiation area and a large internal space in the box, which ensures the interpretation effect of the low-frequency part in the design.
The cabinet of T200a is extremely heavy, and it feels twice as heavy as multimedia speakers of the same size. The cabinet is made of high-density board with a thickness of about 15mm, instead of the composite density board used in ordinary multimedia speakers. The cabinet manufacturing avoids the possibility of resonance caused by sound waves. In addition, there are a lot of attracting cotton inside the cabinet, which artificially increases the damping environment of the sound inside the cabinet, reduces the reverberation time of the cabinet, makes the mid-low frequency part cleaner, and makes the output of the mid-low frequency not feel dragging.
T200a has done a good job in sealing the speakers and the back panel. After removing the tweeter and midrange speakers, we can see that they are both equipped with rubber gaskets to strengthen the tightness between the speakers and the cabinet, eliminating the possibility of leakage. The designers of T200a have independently enclosed the tweeter in a small space, which cannot be directly connected to the cabinet. This avoids a lot of reflections of high-frequency sounds above 2kHz in the cabinet, ensuring the relatively clean sound quality of the mid- and low-frequency parts.
It is not difficult to see from the overall diagram of the back panel (see Figure 4) that the circuit part is composed of four parts from top to bottom: the front-stage operational amplifier, the post-stage power amplifier/rectifier filter, the transformer and the AC input.
The transformer used in T200a is a 75W, ordinary E-type transformer with dual +15V output terminals (see Figure 5). The single voltage +15V power supply is used in the whole circuit to avoid the asymmetry of the positive and negative power supply parameters and destroy the symmetry of the circuit, and can better achieve full balanced amplification. Generally speaking, positive and negative power supplies will have better performance in dynamics. The filter capacitors used in T200a are two electrolytic capacitors with a withstand voltage of 25V and a capacity of 10,000 microfarads (see Figure 6). This capacity of capacitor has never been seen on any multimedia speaker before. Generally, it is good for a multimedia speaker to use a filter capacitor of 4700 microfarads. After filtering, a rectifier tube is used for rectification, and then a part of the power is sent to the upper op amp circuit to provide power.
The circuit design of the power amplifier of T200a still follows the traditional clichés. The op amp stage still uses the ordinary 5532 (see Figure 7).
The volume control of the op amp part of T200a uses a common rotary carbon film potentiometer, and the casing is grounded, which can suppress some noise interference, which is commendable.
The power amplifier of T200a is driven by two LM3886TF bridged and mutually pulled. LM3886TF is a 68W mono high-performance power amplifier. Each LM3886 can provide 38W of power when Vcc is positive or negative 28V, but the power output of +15V voltage provided by the power supply is only about 20W. After mutual pulling, the output power can reach 40W in theory, which is naturally enough for the nominal 75W power of the transformer power supply. It can be seen that the designer has reserved enough power reserve in the design to prevent clipping distortion and overload distortion in large dynamics. However, insufficient Vcc voltage cannot make LM3886 work in the best state, which will more or less have some negative impact on the performance of the power amplifier.
As for the peripheral components of LM3886, it is recommended to use imported tantalum capacitors, CBB metallized non-inductive capacitors, five-color ring precision metal film resistors and other professional luxury-grade components, and use heat sinks to assist in heat dissipation. Here, T200a uses a considerable number of metallized non-inductive capacitors and metal film resistors. However, there are also a large number of ceramic capacitors, and there is only one polypropylene capacitor. Therefore, in terms of the selection of peripheral components, T200a also has a lot of room for enthusiasts to polish.
The frequency division part of T200a adopts the traditional LC circuit. The frequency division circuit of each unit is composed of one capacitor, one inductor and one resistor. The two frequency division circuits of the tweeter and the midrange unit are independent of each other.
Sound quality evaluation
After more than ten days of burn-in and audition, we have come to some conclusions. Overall, if we evaluate the T200a in the field of multimedia speakers, its sound quality performance is unprecedented. In the five-year history of multimedia speakers, no product has surpassed it, and even the M200 from two years ago is far behind it.
The sound quality and tone characteristics of T200a are just like its design positioning as a monitoring speaker. It is neither hot nor cold, and no special effects are incorporated to modify and carve the original color of the sound source. After running-in, the treble of T200a has been very pleasant to the ear, and there is no occasional metallic sound and burrs. The midrange is delicate and round, and the bass is solid and steady. In short, the uniformity of each range is so smooth and natural that it is almost impossible to find peaks and valleys. Its diving depth in the low-frequency part can make people feel insufficient. Some low-frequency details that can be heard with high-end 2.1 speakers are simplified a lot under the interpretation of T200a. However, the low frequency of T200a is very clean, without the feeling of chaos and turbidity. The resolution of T200a is still quite good, but the dynamics are not as good as expected. This is also restricted by the power output that does not put the power amplifier circuit in the best working state.
In short, this T200a is a product for high-end individual users and professional audio producers. It can also be regarded as an entry-level "professional-grade" monitoring speaker. As for its sound quality and timbre and the quoted price of 2,360 yuan, is it worth it? Only after users listen to it can they draw this conclusion.
The most famous speaker unit manufacturers are Dynaudio, Scan-speak, Focal, Audax, Seas, Morel, ETON and JBL. In addition, KEF, Diatone, B & W, Altec, LPG, Visaton, Thiel, Elac, MB and other manufacturers also produce excellent units.
Previous article：JUKEBOX Hard Drive Player DIY
Next article：Pairing with difficult-to-drive speakers
Recommended ReadingLatest update time:2024-11-16 19:41
imx93-audio-buildroot
开关电源设计与制作基础 (蔡宣三，倪本来编著)
- High signal-to-noise ratio MEMS microphone drives artificial intelligence interactionAuthor: Dr. Gunar Lorenz Senior Director of Technology Marketing, Infineon Technologies Proofreader: Ding Yue Chief Engineer, Greater China, Consumer, Computing and Communications Business, Infineon Technologies Introduction At Infineon, I ...
- Advantages of using a differential-to-single-ended RF amplifier in a transmit signal chain designTraditional radio frequency (RF) transmit signal chains typically use a digital-to-analog converter (DAC) to generate a baseband signal. This signal is then up-converted to the desired RF frequency using an RF mixer and a local oscillator. ...
- ON Semiconductor CEO Appears at Munich Electronica Show and Launches Treo PlatformDuring Electronica, ON Semiconductor CEO Hassane El-Khoury was interviewed by Power Electronics News at the exhibition site and gave a detailed introduction to the new products that ON Semiconductor brought to the market. ...
- ON Semiconductor Launches Industry-Leading Analog and Mixed-Signal PlatformThe Treo platform uses a modular architecture that accelerates the development of intelligent power management, sensor interface and communication solutions. The Treo platform is based on 65-nanometer BCD process technology and supports ...
- Analog Devices ADAQ7767-1 μModule DAQ Solution for Rapid Development of Precision Data Acquisition Systems Now Available at MouserMouser Now Stocking Analog Devices ADAQ7767-1 μModule DAQ Solution for Rapid Development of Precision Data Acquisition Systems November 6, 2024 – Offering a broad range of semiconductor and electronics ...
- Domestic high-precision, high-speed ADC chips are on the rise
- Microcontrollers that combine Hi-Fi, intelligence and USB multi-channel features – ushering in a new era of digital audio
- Using capacitive PGA, Naxin Micro launches high-precision multi-channel 24/16-bit Δ-Σ ADC
- Fully Differential Amplifier Provides High Voltage, Low Noise Signals for Precision Data Acquisition Signal Chain
- Innolux's intelligent steer-by-wire solution makes cars smarter and safer
- 8051 MCU - Parity Check
- How to efficiently balance the sensitivity of tactile sensing interfaces
- What should I do if the servo motor shakes? What causes the servo motor to shake quickly?
- 【Brushless Motor】Analysis of three-phase BLDC motor and sharing of two popular development boards
- Midea Industrial Technology's subsidiaries Clou Electronics and Hekang New Energy jointly appeared at the Munich Battery Energy Storage Exhibition and Solar Energy Exhibition
- Guoxin Sichen | Application of ferroelectric memory PB85RS2MC in power battery management, with a capacity of 2M
- Analysis of common faults of frequency converter
- In a head-on competition with Qualcomm, what kind of cockpit products has Intel come up with?
- Dalian Rongke's all-vanadium liquid flow battery energy storage equipment industrialization project has entered the sprint stage before production
- Allegro MicroSystems Introduces Advanced Magnetic and Inductive Position Sensing Solutions at Electronica 2024
- Car key in the left hand, liveness detection radar in the right hand, UWB is imperative for cars!
- After a decade of rapid development, domestic CIS has entered the market
- Aegis Dagger Battery + Thor EM-i Super Hybrid, Geely New Energy has thrown out two "king bombs"
- A brief discussion on functional safety - fault, error, and failure
- In the smart car 2.0 cycle, these core industry chains are facing major opportunities!
- The United States and Japan are developing new batteries. CATL faces challenges? How should China's new energy battery industry respond?
- Murata launches high-precision 6-axis inertial sensor for automobiles
- Ford patents pre-charge alarm to help save costs and respond to emergencies
- New real-time microcontroller system from Texas Instruments enables smarter processing in automotive and industrial applications
- Problems with frequency modulation using MAX2607
- Recruiting BLE software development engineers (with favorable treatment)
- Eliminating Software Failures with MSP432
- Op amp adder and subtractor
- Award-winning lecture: Nexperia Micro Classroom - Explanation of parameters in the power GaN device data sheet
- [Environmental Expert's Smart Watch] Part 12: Configuration of watch name and time
- [TI recommended course] #Boost and buck-boost DCDC converters help wireless charging design#
- Why does the servo motor make noise during operation?
- How to choose LoRa products
- [2022 Digi-Key Innovation Design Competition] Pi-hole Magical Uses of Pi400
京公网安备 11010802033920号