A more modern multichannel surround sound format called Dolby Digital was created in 1992. Dolby Digital, also known as AC-3, overcomes the shortcomings of Dolby Surround. This allowed us to take the sound experience in the home theater to a new level.
As its name implies, Dolby Digital is a digital format, that is, it means recording audio information as a sequence of zeros and ones. Therefore, Dolby Digital is only applicable when using media that record or transmit digital information, such as DVDs, laser discs, or digital satellite television systems. The Dolby Digital soundtrack cannot be recorded on a regular VHS magnetic tape (however, as an experiment, it was recorded on S-VHS). However, Dolby Digital was chosen as the surround sound format for North American DVD releases, as well as the standard for high-definition television. Dolby Digital can also be transmitted using a digital satellite television system, although in this case you will need a special Dolby Digital receiver to receive the audio signal.
In addition to the digital method of encoding information, Dolby Digital is distinguished by the fact that this format provides six discrete (separate) audio channels. Their discreteness eliminates the possibility of unwanted leakage of sound from one channel to another. It should be recalled that the older Dolby Surround system was matrixed: it used folding information from the front and rear channels into two channels - right and left - and then separating them when playing sound. Because Dolby Digital does not mix the different channels, it can provide almost perfect separation. As a result, Dolby Digital has never-before-seen capabilities to precisely localize sound anywhere in your living room.
In addition, Dolby Digital uses a “split” surround channel rather than a mono channel. This means that the left and right speakers can produce completely independent signals. Dolby Digital best conveys the filmmakers' vision.
When using Dolby Digital, all five channels transmit the full frequency spectrum of audio signals perceived by human hearing - from the lowest (20 Hz) to the highest (20 kHz). Another advantage of Dolby Digital is the presence of a completely independent channel for transmitting additional low-frequency information, which allows for the best transmission of sounds arising from explosions and impacts.
Since five discrete wideband channels are used plus a separate LFE channel, Dolby Digital is defined as a 5.1 channel format. In this case, 1 after the point (.1) denotes the LFE channel, which has a limited frequency band.
How Dolby Digital works
Dolby Digital creates a single data stream in which each channel's signals are arranged one behind the other, like six cars on a train. The digital bit rate of all six channels is 384,000 bits per second (384 kbps). For comparison, two channels of digital audio on a CD form a data stream of 1,411,200 bps. In other words, With Dolby Digital, each channel is encoded using ten times fewer bits than a regular CD, and as a result of losing so much information, Dolby Digital produces sound quality that is inferior to CD.
Disadvantages of Dolby Digital Format
The disadvantages of the format include a high degree of information compression (11:1).
Dolby Digital EX
Dolby Digital EX - a format with an additional central rear channel obtained using the matrix encoding principle; denoted as 6.1.
Q. What is DTS?
A. The most accurate surround sound technology available. DTS Digital Surround is an encoding/decoding system that provides six (5.1) 20-bit master quality audio channels. During the encoding process, the DTS algorithm records 6 channels of 20-bit digital audio information in the space previously occupied by only 2 channels of 16-bit linear PCM (regular CD format). During playback, the DTS decoder restores the original 6 20-bit channels. Each of these 6 channels surpasses the 16-bit linear PCM format of conventional CDs in sound quality.
Q. How does DTS compare to Dolby Digital (AC-3)?
A. The DTS master quality format for Laser Disc (LD), Compact Disc (CD) and DVD uses four times less compression than Dolby Digital and results in superior audio quality.
Q. What equipment is needed to play DTS?
A. 5.1 channel electronics with DTS decoder and 5.1 acoustics, that is, a system with 6 separate electronic channels and acoustic systems: Front left and right, rear left and right, center channel and ".1" low-frequency channel for special effects. The electronics can consist of any 5.1 channel surround processor with built-in DTS decoding circuitry or an external DTS decoder, plus 6 channels of amplifier (or 5 channels and a powered subwoofer). Since the introduction of Motorola's advanced chip, which can be easily integrated into any multi-channel device, all new models (including integrated receivers) can contain both DTS and Dolby Digital.
A popular analogy is AM and FM radio broadcasts. You cannot receive AM signals on an FM receiver and vice versa. However, for greater choice, your radio may be equipped with both technologies, so you can select either one with the flick of a switch. And while low-quality AM broadcasting has always found practical applications, the success of FM broadcasting is a clear sign that there is a need for high-quality sound.
Q.What is RF demodulator?
A. In cases where the audio on Laser Video Discs (LD) is recorded in Dolby Digital format, an additional RF demodulator device (built into the decoder or external) is required so that the decoder receives a digital signal ready for decoding. This problem arose due to the fact that the developers had to “squeeze” the Dolby Digital format into the LD audio track format, approved long before Dolby Digital. This problem does not exist for DVD video discs.
Q.SP/DIF specification.
A. SP/DIF - (Sony and Philips Digital Interconnect Format) A digital format approved by IEC958 1989-03 that transmits left and right channel audio information in serial code at speeds up to 48 Kbps with word sizes up to 24 bits. SP-DIF receivers (usually DACs) automatically adjust to the baud rate and bit depth. Electrically, it is a 0.5 volt signal transmitted over a 75 ohm RCA coaxial cable, or a Toslink optical cable. The output of a regular CD player fits into the SP/DIF standard. More details (links) http://www.hut.fi/Misc/Electronics/docs/audio/spdif.html or http://www.cl.cam.ac.uk/Research/SRG/HAN/docs/sp-dif.txt)
Q. Is it true that people actually hear all these differences?
A. At least they are sure of it.
Q. Why do listeners differ in their opinions on sound quality?
A. There are at least three approaches to assessing what “Perfect Sound” is and most listeners place themselves in one of the following three groups:
1.
"It should sound like real live music." These people know how a performer’s live voice sounds, how this or that instrument sounds without an amplifier, they remember how an orchestra sounds in the orchestra pit. Such people want absolutely accurate reproduction.
2.
“It should sound the way the sound engineer intended it.” The sound engineer listened to the sounds coming from the microphones using exceptionally good equipment and mixed them as he considered correct. This may not be the case in reality, but it is exactly how he wanted it to be better.
3.
"This should give me the most pleasure." It doesn't matter whether it sounds good or bad, it doesn't matter what the sound engineer wanted, but if I enjoy listening to some audio equipment, then it's the best.
Based on this, it is clear that no system will satisfy everyone at once. Some want to hear deep bass, others want men's voices to sound like men's voices, others want violins to sound like violins. Audio systems rarely do everything equally well. This is especially true for acoustics. It will be absolutely correct to choose speakers according to your personal taste. Getting used to the sound of the equipment you already have is of great importance. Another reason why most people can't tell the difference in sound: Until you listen to a system that can accurately reconstruct the surround sound space of a stereo signal, you may never realize it's possible. Some recordings of very good quality, when played back on good equipment, make it possible to clearly localize the location of a particular instrument. However, this may not be heard on insufficiently good equipment or recording. Finally, some people simply cannot hear the difference. Not because they are deaf, but simply have an untrained ear or do not know what exactly to listen to, perhaps they have some kind of hearing impairment (for example, age-related decrease in sensitivity to high frequencies or hearing damage from very loud music at rock concerts, etc. ).
Q. Is it true that the type of speaker cables affects the sound?
A. First, the terms: cables are wires with connectors attached to them. The wires are usually multi-stranded, enclosed in a braided screen and insulating material. Cables can introduce noise into the transmitted signal, act as a filter (and thus change the frequency response of the system), and introduce clicks due to improperly connected connectors. It is certain that different cables “sound” differently due to differences in ohmic resistance, interwire capacitance, and different connectors. The effect of exotic twisting of conductors has not been sufficiently studied. In general, these effects (other than resistance) should not be noticeable, although many listeners report significant differences in sound between cables. This effect is specific to each specific system and the only way to achieve the best sound is through trial and error. The connectors must be attached to the column with great force, because... The input impedance of the speakers itself is very low and the transition resistance of the connector of only 1 ohm will significantly affect the power delivered by the speaker.
Q. What types of speaker cables are there and how are they different?
A. There is a wide range of speaker wires available, ranging from $0.2 to $300 per foot. The materials used range from copper to silver. Oxygen-free copper is most likely no different in sound from plain copper, and if you hear a difference, it is for some other reason. Cable resistance is a very important characteristic. The higher the resistance, the greater the effect the cable has on the sound. The conductivity of copper is 1.7 microOhm/cm, silver - 1.6 microOhm/cm, gold - 2.4. Gold does not oxidize, so the contacts do not need to be cleaned periodically, silver does oxidize, but its oxide film is also highly conductive. Copper oxidizes and its oxide is an insulator, so it is not suitable for connectors. The shorter and thicker the cable, the lower its resistance, capacitance, inductance, and the less distortion it causes.
Q. What are the best speaker connectors?
A. The worst connectors are spring contacts. Screw clamps are much better. Gold-plated "slotted blades" or spikes are inexpensive by audiophile standards with very stable mechanical contact and minimal resistance. Gold-plated bananas are also very good for speakers, and although more expensive than spikes, they provide a larger contact area and ease of system reconfiguration. Gold-plated connectors are always better than connectors with any other surface for two reasons: firstly, gold does not corrode even when exposed to active chemicals, and secondly, it is a very soft metal and, when tightly compressed between two plates, it slightly fills There are always scratches and cracks, increasing the contact area.
Q. What about interconnect connections, such as between a tuner and an amplifier?
A. Interconnects transmit a significantly weaker signal than speaker cables. Here the voltage range is from -2V to +2V with a current of several microamps (for comparison, in speaker cables the voltage is from -70V to +70V with a current of several amperes). Interconnects can be regular RCA (unbalanced) or balanced. Most home audio equipment uses regular RCA connections. Better gold plated ones, like Vampire Wire. Unbalanced cables vary in shape, materials and price. Cheap ones have one central core and a screen braid. Medium ones (in terms of price) have a central core of two conductors twisted into a twisted pair and braided. Expensive ones can be of various shapes and from different materials (for example, stranded silver wire with rubber filler). Balanced connections in audio systems are used to eliminate low-frequency hum and other interference caused by power supplies. On unbalanced lines, low-frequency hum is caused by ungrounded source and receiver or differences between the ground potentials of system components with different power supplies. Balanced connections help avoid interference because transmit 2 signals that are out of phase. Isolation can be performed on active elements or on a transformer. Both methods have their advantages and disadvantages. The cables have an XLR connector giving a good mechanical connection. Balanced cables have three conductors: two for the signal and one for the shield. They should be used for lengths of more than 4 meters, when installing professional equipment on judges. For most systems, the most important aspect is mechanical reliability between connector and wire. There are many objective reasons why cables can distort the signal. These are the same resistance and interwire capacitance, the quality of shielding, the quality of the twisted pair in a balanced cable.
Q. Is there any difference between digital cables?
A. There are three types of digital cables from the transport to the DAC or sound processor: coaxial, optical plastic (Toslink) and fiber optic (AT&T ST). Theoretically, they should all “sound” exactly the same (bit for bit). However, there are people who claim to hear a difference, and most of them prefer fiber optics.
Q. What should you look for when buying acoustics?
A. The most important thing is to listen to things on it that are familiar to you. The seller will offer you recordings that will highlight the good aspects of the speaker. So don’t be lazy to take a few of your CDs with you. Don't waste your time switching between two dozen things every 3 seconds. If your budget allows you to choose, ask the dealer to select a couple of different speakers that suit the size of the room and your tastes and listen to each one. Once you've settled on your choice, spend another half hour listening to make sure you're getting your money's worth. It’s very good to just listen to a recording of human speech. Most people can tell when a human voice sounds unnatural. When listening to a musical instrument solo or in a small ensemble, make sure that the sound from it is as it should be. Most people have heard a piano live, so recording a piano is very revealing. Blues, jazz or easy listening music with simple instruments and female vocals is also a good test for acoustics. Try something simple and quiet to hear possible noises, and complex, with many instruments entering at the same time, to ensure that the system works correctly under heavy loads. Sellers usually offer to listen to things that always sound great. Ask to play your favorite recording, so that as a result you can say to yourself: “But I haven’t heard this instrument before.”
Q. What should I look for when comparing speaker systems?
A. When you compare the sound of two different pairs of speakers, try to carefully set the volume on both to be exactly the same. It won't necessarily be at one volume control position. The ear can detect a difference of even 0.5 dB. First and foremost, the sound should be natural. If you are listening to vocals, close your eyes and imagine that the performer is in the same room with you. Sounds natural? The same goes for tools. They should sound the same as in reality. Your very first impression should be: “what a wonderful sound.” If your reaction is: “So many details!”, then most likely these speakers will soon begin to irritate you. If you immediately notice a powerful bass, then the system is probably overweighted with lows. The most common mistake beginners make is buying speakers with very powerful bass because their sound is “impressive.” After a short time, you will get tired of the “blows to the head.” Both low and high frequencies are very important, but nothing should stand out or dominate the listening experience. Sit and listen. Try to hear each instrument separately and the entire piece as a whole, made up of separate parts, so that you don’t get the feeling that some instrument is drawing your attention to itself. Check how the thing sounds at high volume and at low volume. Some speakers “play well” if the sound is quiet and “choke” at high volumes. Others, on the contrary, “wheeze” at very low volumes and work well if the sound is loud. It is acceptable, and sometimes even desirable, to switch from stereo to mono to assess naturalness. Mono is a good test for both room and speakers. The sound image must be stable in the center and not move depending on the volume or frequency of the signal. If mono is not maintained, then stereo is unlikely to turn out good. Large speakers can reproduce low frequencies at high volumes, but this does not mean that small speakers cannot play bass. They just can't do it loud. Good speakers can "restore the stage" by placing some instruments on the left, others in the center, and others on the right. On bad speakers it is much more difficult to determine the positions of the instruments.
Q. Why do you need a subwoofer? How many of them do you need - one or two?
A. Firstly, it is needed to add bass to systems where there is not enough bass. And secondly, transfer low frequencies to a separate loudspeaker and thereby reduce the level of intermodulation distortion caused by nonlinear mixing of signals of different frequencies. Thirdly, a subwoofer will increase the overall power of your speaker system. To improve the sound of a good system, the subwoofer should be built into it "softly" without bursts or dips in the frequency response, expanding the already existing lower range. Most subwoofers have a crossover, which is located between the amplifier and the speakers and divides the lower frequencies into the subwoofer and the higher frequencies into the speaker. This separation may improve the system, may not affect the sound, or may harm it. Most small speakers have a low-frequency boost to compensate for the performance penalty of their small size. A well-made subwoofer will improve the sound of small speakers. Even large speaker systems can benefit from a subwoofer if configured correctly. However, the better the original system, the more likely it is that an additional subwoofer will cause damage. To the human ear, low frequencies are not determined by direction, so some might say that one subwoofer is enough. This is correct, but not entirely. Some people believe that you need two subwoofers to properly reproduce a stereo image, even though there is little stereo information in the low frequencies. Another reason is that in addition to very low frequencies, the subwoofer reproduces frequencies up to 100Hz, and this is already a frequency with a discernible direction, if, of course, the subwoofers are located at least a meter from each other. Finally, although musical instrument recordings rarely contain frequencies below 50Hz, there is almost always low-frequency acoustic noise that can add a sense of spatiality if subwoofers are spaced far apart.
Q. How do I connect a subwoofer to a stereo system?
A. Many subwoofers (powered) have their own amplifier and crossover. These are connected directly to the preamplifier output. Passive subwoofers can be connected in parallel to the main speakers or a separate bass amplifier with crossover. Refer to the connection guide included in the package.
Q. What is a receiver?
A. This is a tuner, amplifier and preamplifier in one package. Typically, the receiver has inputs for a CD player, cassette deck, vinyl turntable and 1-2 additional inputs. It has an input switch, volume and tone controls, as well as outputs for a pair or more speakers.
Q. What is a tuner?
A. A tuner is a radio receiver that cannot be connected to speakers directly. Sometimes the receiver in the tuner is of higher quality than the receiver. All tuners accept the FM band, and some also accept AM. The signal from the tuner needs to be amplified.
Q. What is DAT format?
A. Today this is the most common professional digital format of 2-track recording on magnetic tape. A DAT (Digital Audio Tape) cassette holds 2 hours of recording with the same resolution and dynamic range as a CD. Recording can be done at 32 kHz, 44.1 kHz (for CD) and 48 kHz (DAT standard). A DAT cassette is similar in appearance to a VHS cassette.
Q. What are Dolby B, C, S, HX Pro and DBX?
A. Dolby B, C, S and DBX are systems that improve the signal-to-noise ratio of magnetic recordings. They all work approximately the same way: they reduce the dynamic range of the signal during recording and expand it during playback. Dolby B works primarily at high frequencies, increasing their amplitude during recording and decreasing their amplitude during playback, while the high-frequency noise of the tape itself is also reduced. A Dolby B recorded cassette can be listened to on a regular tape recorder without Dolby B processing, but the sound will be too bright, which will have to be compensated by the tone control. With Dolby C processing, a greater (8-10 dB) noise reduction is achieved than Dolby B due to the use of a larger frequency range and volume change level. Playing such a cassette on a regular tape recorder without Dolby C sounds quite unpleasant to most people. Some people believe that tapes recorded in Dolby C only sound best on the deck on which they were recorded. Dolby S operates over an even larger frequency range than Dolby C and achieves even greater noise reduction. Advantages of this system: 1. Many people believe that using Dolby S gives a closer proximity to the original than Dolby C, 2. Dolby S recordings do not sound as terrible when played back on Dolby B decks, 3. The Dolby S recording tape is not as sensitive to change deck type. DBX is similar to Dolby B, C and S, but uses compression at all frequencies - low and high. but uses the same compression on all frequencies, high and low. However, DBX is more common in the professional market. DBX is not compatible with Dolby and cannot be played back on a regular tape recorder. All noise reduction systems have two disadvantages. One of them is that the compressor will not be able to suppress a strong signal until it “hears” at least a short fragment of it. It is this fragment that passes to the output without compression. The same applies to a weak signal, which remains unrecovered for a certain period of time. These delays are clearly audible when listening. The second drawback is that if there are flaws in the frequency response of the system, then the compressor will only amplify them. For example, if there is a 2 dB dip in the frequency response of a tape recorder at a frequency of 1 kHz, then during playback there will be a 4 dB dip in the signal. Therefore, many people prefer a system without noise reduction at all. Dolby HX Pro has nothing to do with noise reduction or compression and is designed to improve frequency response when working with cheap tapes.
Q. Can CrO2 tape damage a deck designed for normal film?
A. Can not. Everything will work fine. These belts are no more abrasive than regular ones. Decks designed for CrO2 or Metal tapes have a different offset and frequency correction in order to “squeeze” the maximum out of the tape and level out the frequency response.
Q. What is the difference between VHS HiFi and Beta HiFi?
A. VHS HiFi audio uses amplitude modulation; in Beta HiFi - frequency.
Q. How do audio recordings on a HiFi VCR, a regular tape recorder, and a DAT tape recorder compare?
A. VHS HiFi and Beta HiFi are analog recordings that use modulation to record video and audio signals onto video tape. VHS tapes have special audio tracks. In terms of HiFi quality, VHS recordings are better than inexpensive cassette decks, but worse than good cassette decks with noise reduction systems, and worse than DAT recordings. Therefore, in cases where you need high-quality recording, use a deck with a manual recording level setting.
Q. Which is better: a stereo system or a system of individual components?
A. Some stereos sound quite good. Even several "audiophile" manufacturers produce such combined systems. At the muses centers have a number of advantages: they are cheaper, take up less space, are easier to manage and install, and there is no web of wires. If you are planning to purchase a music center, then it is better to do this in a hi-fi store, where you can “listen” to the purchase and choose products from audiophile companies, for example TEAC. The weakest point of a good music center is its speakers. The best music centers are generally supplied without speakers, leaving the buyer the opportunity to choose according to his taste. Why then is there so much talk about systems of individual components? You can get significantly better sound from a multi-component system. A component system is more difficult to install, configure, takes up more space and is more expensive. At the same time, such a system has flexibility. You can replace the power amplifier or CD player if you are not satisfied with them. You can select individual components. In the case of music The center cannot do anything if you are not satisfied with something. To most muses. centers you cannot connect an MD, a vinyl player, a VCR, or many other necessary equipment.
Q. What is bi-amping? bi-wiring?
A. Most speaker systems are connected to the amplifier with one pair of wires for each speaker. Inside the speaker, the signal (pre-converted) is distributed to the speakers by a crossover. Some speakers have the ability to connect with a bi-wiring cable, in which 2 pairs of wires are routed from one amplifier output to the speaker; one pair connects to the speakers serving the mid and high frequencies, and the second pair connects to the woofers. Buy-wiring (that's how it's pronounced) is treated differently. Some claim they don't hear any difference, others claim they hear a noticeable difference. The most acceptable explanation may be that separating the cables also separates the noise induced into them (mid and high frequencies that are weak in power and low frequencies that represent the main load for the amplifier). The cables therefore need to be spaced apart from each other by at least several. centimeters. In all cases the effect appears to be weak. Biamping means that each speaker that allows bi-wiring has 1 stereo amplifier connected, one channel of which is connected to the HF connectors, and the second channel is connected to the LF connectors of this speaker. Accordingly, the second speaker is connected through its own stereo amplifier. In another configuration: one stereo amplifier is loaded with the high-frequency speakers of both speakers, the second, respectively, with the low-frequency speakers. The meaning of by-amping (pronounced this way) is that significantly more power is needed to “drive” the low-frequency speakers than to drive the high-frequency speakers. In this way, you can connect the low ones to a powerful stationary amplifier, and for the high ones you can choose an amplifier of better quality, but weaker.
Q. What do the letters "ADD" on my CD mean?
A. This letter code has nothing to do with the quality of the recording. It refers to the equipment and instruments used in recording. The first letter A means that the original recording was in analog form (D - in digital). The second letter indicates what equipment was used to mix and edit the recording. The third letter indicates the equipment used in the final recording; for a CD, of course, it is always “D”.
Q. What is Class A, Class B, Class AB amplifier. What are class C and D?
A. All of these terms refer to the characteristics of the amplifier output stages. In short: Class A amplifiers sound the best, cost the most, and use the most electricity; Class AB amplifiers rival Class A in sound quality, dominate the market, use less power, are cheaper, smaller and lighter. Class D amplifiers are used only for special applications, e.g. bass guitar or subwoofer amplifier. They are even smaller than class AB, more efficient, and amplify signals below 10 kHz (less than the audible range). Classes B and C are not used in audio amplifiers. Class A includes amplifiers with output transistors that are constantly biased, so that current constantly flows through them. Their most important advantage is that their frequency response is the most linear, i.e. introduced distortions are minimal. The main disadvantage is inefficiency, because... even in silent mode, the amplifier continues to heat the atmosphere. A 50-watt Class A amplifier is large and heavy. Among high-end amplifiers, only 10% operate in class A. In class B amplifiers, the output transistors have zero bias, so at zero signal the transistors are off and do not conduct current, but introduce very strong distortion due to the nonlinearity of the gain around zero. These distortions are very audible and make such amplifiers unsuitable even for unassuming listeners. Class C amplifiers are similar to Class B with all its disadvantages and also have no application in hi-fi. In class AB amplifiers, there are 2 transistors in the output stage, as in class B, but both with a slight bias, which brings both transistors to linear mode. Most amplifiers on the market are class AB. Class D amplifiers use pulse modulation in the output stage, i.e. transistors are either open or closed. Therefore, such amplifiers are the most economical and efficient. some reach 80%, but the distortion introduced by such amplifiers is slightly higher than that of class AB. At the output of a class D amplifier there must be a passive low-pass filter to get rid of the switching frequency. This filter introduces both phase shift and nonlinearity, and also cuts high frequencies, making it beneficial only for subwoofers. It is, of course, possible to make a class D amplifier with a full frequency range, but in this case you will have to significantly increase the switching frequency of transistors (>>40 kHz), build a high-quality non-distorting filter, however, in the rest of the devices of your system, the increased switching frequency will be successfully induced in the form of interference .
Q. What is Super Audio CD?
A. This is a new standard (abbreviated as SACD) Promoted by: Sony, Philips. Supported: Mobile Fidelity, DMP, Telarc. Of course, the music catalogs of Columbia and Sony are also candidates for release in this format. Official introduction is planned: in Japan in the spring, in America - in the fall of 1999. Double-layer CD. The outer translucent layer (HD) contains 2-channel or multi-channel high-resolution music recorded using the Sony process DSD (Direct Stream Digital). The high capacity of this layer is achieved by reducing and compacting the pits. The inner layer (CD) is a regular CD layer with a resolution of 16 bits/44.1 kHz. Full compatibility with current CD players. It is supposed to release new disc titles simply with the SACD mark, which the buyer may not pay attention to. After some time, he has a collection of SACD recordings and is ready to upgrade his player. from a technical point of view, full CD compatibility is questionable. Will all CD players be able to read the inner layer through the outer layer normally? Further, the cost of disc production is inevitable, although Sony does not disclose specific figures. It turns out that the owner of a CD player, satisfied with 16/44.1 sound, will be forced to pay more for new discs, which will sound worse on his device than the old ones?
Q.Tell us about DAD disks.
A. The so-called DAD discs promoted by Classic Records use the DVD video standard, allowing 2-channel 24-bit/96 kHz audio to be recorded onto DVD. The upcoming DVD audio format makes this temporary measure unnecessary.
Q. What are the prospects for a purely audio format for DVD?
A. The DVD-AUDIO standard is being promoted by: JVC, Matsushita, Pioneer, Toshiba. Supported by: Warner Music Group. Scheduled official introduction: fall 1999 (difficulties with copyright enforcement remain). Here are its main properties:
1.
Quantization frequency, word length (number of bits in each sample), number of channels and other parameters are determined by the recording producer. For example, a disc might contain a 2-channel 24/96 mix and simultaneously a 6-channel version of the same music with 24/96 front channels and 16/48 rear channels. Possible sampling frequencies are 44.1 kHz, 48 kHz, 88.2 kHz, 96 kHz, 176.4 kHz and 192 kHz with any resolution from 16 to 24 bits. Recordings encoded at 176.4 and 192 kHz are limited to two channels. Transmitting a multi-channel signal in this case would require exceeding DVD's maximum bit rate of 9.6 million bits per second. The recording producer selects recording parameters depending on the characteristics of the music, the duration of the fragment, and the quality of the recording equipment. The recording parameters are transparent to the user: he simply presses the PLAY key, and the player determines them independently.
2.
DVD audio players will be equipped converter making 2-channel mix from multi-channel recordings. The record producer can define the sound of this mix by inserting codes into the data stream that control the player's converter. This feature is called SMART (System Managed Audio Resource Technique) Content. If the producer has not inserted these codes, the mixing occurs randomly. As mentioned above, a producer can record his own 2-channel version on disc in addition to the multi-channel version, but this requires additional disk space, and therefore leads to a decrease in either the playing time, or the quantization frequency, or the resolution.
3.
Another option for using SMART Content: the producer can put it there player or decoder settings, which he considers necessary for the optimal sound of the recording.
4. MLP (Meridian Loseless Packing)
Even a DVD audio disc cannot accommodate 6 channels of digital audio quantized at 96 kHz with 24-bit resolution. Therefore compression is necessary. In 1998, MLP became part of the DVD audio standard. All DVD audio players will contain an MLP decoder, but MLP is not required for audio recording. If the disc is short or only 2 channels are used, the recording producer may not use MLP. MLP provides disk space savings of approximately 40% compared to unpacked volume (see table). Importantly, MLP is truly a lossless compression process: after packing and decompressing, the data is absolutely (bit-by-bit) identical to the original data, and the audio quality is the same. This gives it a distinct advantage over lossy compression methods such as Dolby Digital and DTS. Since DVD-ROM drives will also play DVD audio discs, each DVD-ROM drive will contain an MLP decoder. That is, a small British high-end company will receive a royalty from each sold personal computer equipped with a DVD-ROM drive.
Q. Will DVD audio players have digital outputs?
A. The availability of high-resolution digital outputs on DVD audio players is an understandable concern for record companies. Therefore, the digital outputs of DVD audio players will not be based on the usual SPDIF, AES/EBU interfaces that we have in CD players. Instead, the digital output will be a multi-pin connector called FireWire or, by its technical name, IEEE1394.
FireWire is a very wideband interface that can carry digital video, high-resolution multi-channel digital audio, computer data, and control codes. These codes allow all the components of a hi-fi or home theater system to "talk" to each other, so that each component "knows" what other components are connected to the system. This bidirectional communication makes it possible to disable the DVD Audio digital outputs under certain circumstances. For example, if you want to play a DVD audio disc using an external DAC, the control codes will allow this. But if you try to record this data stream, the FireWire interface will tell the DVD audio player that the digital output is connected to the recorder and will disable the digital output.
It looks like DVD audio players won't have digital outputs until the FireWire standard becomes widespread. Instead, look for 6 RCA analog outputs. High-end companies will be forced to release single-block DVD audio players with six DAC channels. However, for stereo purists there are still several 2-channel DVD audio devices.
(AC-3, ATSC A/52) (Dolby Digital) - a spatial sound reproduction system developed by Dolby Laboratories, Inc. (“Dolby Labs”), led by Ray Dolby, a pioneer in the audio and video industry.
The format is standardized by the Advanced Television Systems Committee, coded A/52, and Dolby Digital (DD) is a trademark.
Modern Dolby Digital systems provide six channels of digital surround sound. Left, center and right front channels allow you to accurately determine the position of the sound source on the screen. Separate "split" left and right rear side channels enhance the sense of presence, creating volume. And an additional low-frequency channel adds intensity to the action on the screen.
In the film industry, the Dolby Digital soundtrack is encoded optically directly onto the film strip in the spaces between the punch holes. Placing a digital soundtrack on the same media as the film allows it to co-exist with the analogue track without the need for additional storage media, and also ensures absolute synchronization of image and sound.
Dolby Digital surround sound system supports 6 channels:
- front left and right
- front center
- rear left and right
- subwoofer
Thus, in total, Dolby Digital is suitable for 5.1 surround sound reproduction.
Data compression
One two-hour movie requires about four gigabytes for the audio track alone, if the audio data is not compressed. A double-layer DVD has a total capacity of about 8 GB. Therefore, audio data compression is very important. At the same time, when compressing audio, losses constantly occur. The AC-3 method is often used for compression. Like other audio compression methods, AC-3 removes sounds that are not perceptible to the human ear, thereby reducing the amount of information stored.
Bitrate in AC-3 reaches from 32 to 640 Kbps. In a movie theater, the Dolby Digital bitrate is 329 Kbps, on DVD with 5.1 audio - 384 or 448 Kbps.
Dolby Digital Technologies
Dolby Digital EX
EX is a prefix used to designate Dolby Digital sound systems with 7.1 channels: two front, center, low-frequency, two rear surround and two front surround.
Dolby Digital Surround-EX
Dolby Digital Surround-EX adds a third surround sound channel to the audio track. The idea belongs to the sound engineers of the Skywalker Sound studio. The technology was developed jointly with Dolby Laboratories and Lucasfilm THX.
Dolby Digital Live
Dolby Digital Live (DDL)- technology for encoding multi-channel (5.1) audio signals into AC3 format in real time, proposed by Dolby Technologies. Designed to transmit multi-channel audio from games and other applications to the receiver via the S/PDIF interface (optical or coaxial). Its use allows you to get rid of the restrictions due to which only ready-made (i.e. stored encoded in the AC3 or DTS format) multi-channel tracks, usually the soundtrack of films, could be transmitted via digital interfaces), and in games the digital output capabilities were limited to the usual stereo sound.
Dolby Digital Plus
MIPS Technologies and Dolby Laboratories have introduced new audio technology for devices that support high-definition video and audio playback, such as HD DVD and Blu-ray players. The audio technology is called Dolby Digital Plus and can be used in MIPS32 microprocessor cores.
Dolby Digital Plus will also improve the quality of recording audio content on HD DVD and Blu-ray Disk media, thanks to the support of an even greater number of channels than was possible with Dolby Digital. The companies will present to developers of SoC solutions (System-on-Chip) varieties of the integrated Dolby Digital Plus codec based on the MIPS core.
Peculiarities:
- Multichannel audio with independent channels
- Supports up to 7.1 channels* and the ability to have multiple audio programs in one stream
- Dolby Digital stream output for compatibility with older devices
- Maximum flow rate up to 6 Mbps
- Bitrate from 3 Mbps on HD DVD and up to 1.7 Mbps on Blu-ray Disc
- HDMI supported
- One stream may contain material in different languages
- New encoding options for audio professionals
- Maintains high quality at more efficient broadcast data rates (200 Kbps for 5.1 channels)
Dolby Digital Plus supports more than 8 audio channels. The HD DVD and Blu-ray Disc standards currently limit this number to 8.
Dolby TrueHD
Dolby TrueHD is one of the first two uncompressed (lossless compressed) audio formats available only for optical HD players. Although the Dolby TrueHD codec is optional, the format is widely supported by Blu-ray players and discs.
Dolby TrueHD uses the Meridian Lossless Packing (MLP) compression algorithm. The Dolby TrueHD digital stream can accommodate up to 14 individual audio channels, but in practice it works with 6 (5.1) or 8 (7.1) channels.
The Dolby TrueHD track specification for Blu-ray movies is as follows:
- Audio codec - Dolby TrueHD.
- The channels (sound scheme) are almost always 5.1, 6.1 and 7.1 are very rare.
- Audio clarity data is often not available, but typical values are: 16 bits at 48 kHz or 24 bits at 48 kHz; for some live discs these values are 24 bits at 96 kHz.
- The stream value is usually not available, but is usually 4608 kbps (4.5 Mbps, which corresponds to six channels at 48 kHz and 16 bits). The highest we've seen on commercial concert Blu-ray discs was 9.0 Mbps, which equates to six channels at 96 kHz and 24 bits. The maximum value for Blu-ray is 18 Mbps.
Also known as:
In 1965, American physicist and engineer Ray Dolby founded Dolby Labs in London. He worked on the idea of creating noise reduction systems that would improve sound quality. The technology had to be suitable for both professionals and the average public. Since then, the name Dolby has become known throughout the world, and the surround sound standards created in the laboratory are used both in cinemas and in the homes of our citizens.
Dolby Digital sound first appeared in theaters in 1992 with the premiere of Batman Returns, and since then has been heard in nearly a thousand films around the world, and is one of the most advanced developments from Dolby Laboratories.
Dolby Digital 5.1 is also called AC-3 (Audio Code-3). The standard provides six channels - two in the front, two in the rear, one in the center and one for the subwoofer. Unlike Dolby Surround And Dolby Prologic, here the frequency bandwidth has been extended from 20 Hz to 20 kHz. 5.1 means five front and rear channels plus a subwoofer. Dolby Digital sound, at the request of the manufacturer, can be recorded in five separate full-frequency channels - left, right, center, right effects channel, left effects channel. In addition, there is an additional channel of powerful low-frequency effects (LFE, low-frequency effects) that are more felt than heard. Since this channel has a bandwidth ten times narrower, the channel is called LFE.1
As its name implies, Dolby Digital is a digital format, that is, it means recording audio information as a sequence of zeros and ones. Dolby Digital differs in that this format provides six discrete (separate) audio channels. Their discreteness eliminates the possibility of unwanted leakage of sound from one channel to another. The subwoofer channel is also called LFE (Low Frequency Effects). The term AC-3 refers to an audio coding technology that discards inaudible audio and distributes the rest into six channels. Accordingly, to play Dolby Digital sound, you need an external decoder or sound card that decodes 5.1 sound and is equipped with the appropriate outputs.
This encoder was designed to take full advantage of the human ability to mask audio by breaking the audio spectrum of each channel into narrow frequency bands of varying sizes optimized for the frequency selectivity of human hearing. This allows the digitization noise to be very precisely filtered so that it is very close in frequency to the frequency components of the desired audio signal. By reducing or even completely eliminating noise where there is no masking audio signal, the sound quality of the original signal is not subjectively affected. In this key aspect, encoding such as AC-3 is a form of highly selective and high-quality noise reduction.
When moving from analogue recording of a signal to recording on a digital medium such as a compact disc, it is discovered that the digital encoding of audio signals used in CDs produces too large amounts of data to be efficiently stored or transmitted electronically, especially in cases where it is necessary encode multiple channels. The result has been new forms of digital audio coding - known collectively as "perceptual coding" - that have been designed to allow low-bitrate data streams to be used with minimal perceptible loss of audio quality. An example of such an encoding algorithm is the third generation of Dolby encoders - AC-3.
Dolby Digital uses audio compression at a fixed ratio of 1:2. In other words, no matter what sound you take, after compression you will get a strictly defined stream. This has its pros (disk space) and cons - since the sound quality still deteriorates. However, as a result, the sound does not take up so much space on the DVD, which allows you to add translations into different languages and various bonuses. AC-3 uses 18-bit encoding, so the output stream of AC-3 is 384 kbps. To restore the sound picture, the Dolby Digital decoder adds a delay of 1 ms to the front channels, since the listener is often closer to the rear speakers rather than the front. The delay allows you to balance the sound picture. Some decoders allow you to change the delay.
Dolby Digital sound can be obtained when playing laser discs, DVD-Video discs, some computer DVD-ROM discs, digital cable and satellite programs, and digital TV broadcasting (DTV). Typically, programs containing Dolby Digital are marked with special symbols.
The Dolby Digital soundtrack cannot be recorded on a regular VHS magnetic tape (however, as an experiment, it was recorded on S-VHS). Dolby Digital has been chosen as the standard for high definition television. Dolby Digital can also be transmitted using a digital satellite television system, although in this case you will need a special Dolby Digital receiver to receive the audio signal.
The main advantage of the standard is that Dolby Digital has become the surround sound standard on DVD. The DVD standard states that the disc cannot use other audio technologies unless it has a separate Dolby Digital track. So you will never find a disc with only DTS. That's why Dolby Digital is universal.
Predecessors of the Dolby Digital standard:
- Dolby Surround- three channels, two for front sound, one for rear sound. The range of transmitted frequencies is from 100 Hz to 7 kHz.
- Dolby Pro Logic- Improved Dolby Surround by increasing the number of channels to four, including the center channel, and using one or two elements for rear sound.
Dolby Digital soundtracks have a very wide dynamic range between quiet and loud passages. At full volume you can hear the vibration providing a true theater experience. However, late at night, loud movie soundtracks may irritate other members of your family or neighbors. But if you turn down the overall volume to "restore peace and quiet," dialogue can become very hard to hear and ambient sound disappears completely. What you'd really want in this situation is to turn down the loud effects, turn up the volume on the quiet passages, and leave the dialogue at the volume. That's exactly what Dolby Digital's dynamic range control system allows you to do. For low-volume listening, it applies dynamic band compression to preserve quiet sounds, prevent overly loud effects, and keep dialogue loud. To make this feature easier to understand, Dolby Digital equipment manufacturers call it Midnight Mode.
The Dolby Digital encoder is capable of processing an input signal with at least a 20-bit dynamic digital signal with a frequency range of 20 to 20,000 Hz ±0.5 dB (-3 dB at 3 and 20,300 Hz).
The low frequency channel covers the range from 20 to 120 Hz ±0.5 dB (-3 dB at 3 and 121 Hz).
Sampling rates at 32, 44.1 and 48 kHz.
The output data stream width ranges from 32 kbps for one mono channel, to a maximum of 640 kbps.
Typical rates are 384 kbps for "5.1" Dolby Digital consumer formats, and 192 kbps for two-channel audio.
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