After using the WRT cable, you can refer to the device sound characteristics introduced in this article to accurately obtain the sound effects you want. Reduce the inconvenience of frequently changing equipment and save your resources.

Audiophiles may unknowingly follow the trajectory set by the market’s big V for us. Here we only introduce the equipment from the technical and physical inevitability, Some content may be different from what you have previously understood.

Sound characteristics of audio equipment

In the audio market, we can find many related introductions of various equipments. Usually, they will describe the various characteristics in the equipment from the positive, and rarely explain the negative effects behind the characteristics. However, these negative factors control the quality on the music playback function. To get the sound effects you expect, you need to fully understand the equipment. In the following content, we will technically analyze what are the deficiencies in the equipment, what impact will it bring to the functional quality, and what sound characteristics will appear from this.

We do not intend to evaluate the quality of the equipment, nor do we intend to impeach the defects of the equipment. Here is just to introduce the sound characteristics of these devices from the perspective of principles, so that when you use these devices, you can give full play to their strengths and avoid weaknesses. If necessary, you can quickly determine the location of the sound effect based on these characteristics. It must be known that, like everything else, audio equipment is not absolutely good or bad. In daily music appreciation, as long as the equipment can meet the sound effects you like, it is a good equipment for your sound system.

Everything in the world should have an explanation, and the sound system is no exception. Audio equipment is a product of electronic technology, and its various phenomena must conform to physical principles and should be explained technically. In principle, all types of equipment have sound characteristics produced by their physical characteristics. Although each device of the same device has its own sound personality, the common characteristics of their device types will still exist.

In the following, we will use the term micro-dynamics. Micro dynamics is the ratio of two voltage (or sound pressure) values in a time interval. In some cases, it is equivalent to what audio enthusiasts call micro-dynamics. The only difference is the length of the time interval. Generally, the minimum time interval for the human ear to confirm the existence of two sound pressure intensities (or there are two sounds) from the sound is 0.1 second. The micro dynamics we are referring to here include very short time intervals and extend to a range that can be distinguished by the human ear.

We will also use the term "Sound State" in the following description. Sound State——The emotional state of the sound. Technical expression: The sound pressure value will change over time. This change relationship is expressed on the coordinate graph of the amplitude vertical axis and the time horizontal axis, which will produce a trajectory of the sound pressure value changing with time. Sound state-the shape of the track of sound pressure changes over time.

In the product introduction, we will find some technical indicators related to the equipment. How will various indicators affect sound quality? How big is its impact? Which indicator has the greatest impact? This is a topic that audio enthusiasts often pay attention to and are discussed from time to time.

The phase characteristics of the sound system will directly affect the sound state. Phase distortion will inevitably affect certain frequency components in music electrical signals, causing them to be misaligned on the time axis. This will change the instantaneous value of sound pressure at certain timing positions and will also produce micro-dynamic distortion in a short time interval.

The various distortions of the sound system, they all change the signal waveform in the end, and specifically change the sound state and timbre.

About timbre
Among various distortions, intermodulation distortion has the greatest impact on timbre. When a music signal passes through an amplifier with intermodulation distortion, some frequency components that are not in the original signal will appear on the signal spectrum, thereby changing the timbre.

Non-linear (harmonic) distortion will also cause some frequency components not in the original signal to appear on the signal spectrum, thereby changing the timbre of the sound. Unlike intermodulation distortion, the number of frequency components produced by nonlinear distortion has a linear relationship with the number of original signal frequency components, but the relationship reflected by intermodulation distortion is exponential (to be precise, it is called mathematically Combination relationship)).

The meaning of the distortion indicator
Each amplifier includes multiple unit circuits. Each unit circuit is also composed of multiple circuit segments. When each section and each unit circuit itself has large distortion, the non-linear distortion value of the amplifier (from input to output) can be reduced through mutual compensation and large loop feedback of each section and each unit circuit. The ideal degree is close to zero. However, these measures have not eliminated the non-linear distortion existing in each segment and each unit circuit itself.

Segmented circuits with nonlinear distortion will inevitably produce intermodulation distortion. A signal containing two or more frequency components, passing through a segmented circuit with nonlinear distortion, will inevitably have additional frequency components that the original signal does not have. Once these intermodulation distortion products appear, they cannot be separated or removed from the signal.

Intermodulation distortion and nonlinear distortion are closely related, but they have different meanings. If the signal processed by the amplifier is a sine wave (only a single frequency component), the nonlinear distortion index has a certain meaning. If this is a music signal (contains two or more frequency components), the index of intermodulation distortion will be able to more fully reflect the overall characteristics of the amplifier.

The technical indicators
For equipment users, the significance of technical indicators is that there can be a unified standard to evaluate and compare the work quality and performance differences of the same functional products of all parties. If the established technical indicators meet the working nature of the equipment, the evaluation is valid. Audio equipment manufactured by modern technology usually has some perfect technical indicators. So, can we effectively judge the quality and characteristics of their voices based on these indicators?

We can discuss this issue with an operational amplifier as an example. If op amps of the same company, the same model, different origins, or even the same model and different batches are installed on a typical audio circuit for testing. For some technical indicators currently used to measure audio equipment, we can all measure the same value. However, when we install these operational amplifiers on the same audio equipment for sound comparison, we will hear different sound effects.

For audio equipment, the above example is a common phenomenon. It shows that these indicators are not enough to fully reflect the actual working quality of audio equipment. Many static indicators are used to measure audio equipment in industrial production. However, in addition to more directly reflecting the phase characteristics and intermodulation distortion of the sound quality of the audio equipment, most of the static indicators will gradually lose their meaning as the signal alternating speed increases and the dynamic increases. Therefore, based on the existing technical indicators, audiophiles may still be unable to effectively evaluate the actual quality of the equipment in actual use.

Similarly, these indicators alone cannot effectively analyze the sound characteristics of audio equipment. Below, we will analyze the inevitability of the sound characteristics of the device based on physical principles (electronic principles obey physical mechanisms).

Please note: When using a WRT cable, the sound characteristics described below can be accurately presented. If the audio system uses other cables, the sound state may be somewhat distorted, and the apparent degree of characteristics will vary.

The quality of grounding has a great influence on the stability of electronic equipment. Electronic equipment transmits or processes electrical signals, and uses "ground" as a reference to establish a basis for comparison. If the "ground" is unstable, the accuracy of the foundation will be shaken. Set up an independent ground for the audio system to avoid external interference. The internal resistance of the grounding will affect the degree of mutual interference between the various components of the audio system.

If we classify amplifiers by device type, we can divide them into electron tubes, transistors, and integrated circuit amplifiers. Affected by the electrical characteristics and physical structure of the device, various devices used in audio (low-frequency linear) circuits can basically have such characteristics: the micro-dynamic filtering* of the tube amplifier is the largest, the transistor is the second, and the integrated circuit is the smallest (but in In some applications, after the integrated circuit amplifier is combined with the peripheral circuit, the strength of the micro-dynamic filtering may be stronger than that of the tube amplifier). The intensity of micro-dynamic filtering controls the density of sound details. With the enhancement of micro-dynamic filtering, the original details of the music will gradually decrease. In the circuit design, the intensity of the micro-motion filter can be controlled to improve the smoothness of the sound, but the direction of control can only be enhancement.

* Low-frequency analog circuits used in audio equipment usually need some measures to filter out noise, overshoot, and high-frequency interference. Regardless of the original intention of adding these measures, they will correspondingly produce some negative effects of micro-dynamic filtering. The weak micro-dynamic filtering only reduces the sound details. When the intensity reaches a certain level and combined with loop feedback, these measures will produce an effect of compressing signal dynamics.

Here, by the way, a frequently heard topic "Amplifier without Feedback". This feedback is of course negative feedback. Because positive feedback is the oscillator. We need to understand a simple concept: linear circuit. Linear circuits are also called analog circuits. The traditional amplifier is an analog amplifier. Now there is a digital amplifier in the audio market, that is, a class D amplifier. Analog amplifiers are also called feedback amplifiers or linear amplifiers. Because no device can have a completely linear amplification characteristic. To achieve linearity, the amplifier must introduce negative feedback. So any linear amplifier must be a feedback amplifier.

Every amplifier design hopes to achieve a balance in the smoothness of the sound, the details of the music, the sense of speed, and the micro dynamics. Because power amplifiers need to drive complex inductive and capacitive loads, it is more difficult to satisfy this balance. Therefore, the design of some pre-amplifiers has increased the consideration of matching certain power amplifier characteristics to achieve this balance in the overall coordination of the system. Using this effectively, we can find many combinations of amplifiers that can be matched with each other.

Earlier preamplifiers have more functions. They have more complicated wiring, making the sound richer, but also accompanied by some muddy. In the period when operational amplifiers were widely used, this muddy sound has become the sound characteristic of audio equipment in this era. However, on some line amplifiers with relatively simple circuit structure, we can still hear a relatively clean sound.

Later produced preamplifiers. As the signal source gradually becomes more single, the circuit structure of many preamplifiers is closer to the line amplifier in design. The circuit structure becomes simpler, combined with the improvement of component quality and technological progress, the timbre of the amplifier can be controlled more artificially. Therefore, there are more preamplifiers with various sound effects for us to choose from.

Some preamplifiers are equipped with tone or working bandwidth control functions. These functions will change the phase characteristics of the amplifier, thereby changing the sound state.

There are some passive volume controllers on the market that are used to replace preamplifiers. The use of such equipment will require: 1) High-quality signal interconnection lines, whose length should be as short as possible, especially the output end interconnection lines; 2) The receiving end equipment needs to have high input resistance and low input capacitance. However, most semiconductor devices do not have high input resistance; most vacuum tube devices do not have low input capacitance. Therefore, it will be more difficult for this type of equipment to get a perfect fit. If the above conditions are not met, after using this type of equipment, the frequency response and phase characteristics of the audio system will undergo considerable changes.

The audio system is connected to an analog equalizer, which will produce severely chaotic phase characteristics. Changes in the phase characteristics will inevitably affect the sound state. The digital equalizer has good phase characteristics. However, the signal requires A/D and D/A conversion in a digital equalizer, and the sound quality depends on the functional quality of these conversion circuits. Digital equalizers with digital inputs and outputs do not have the above problems. Many new generation professional mixers use this type.

Power amplifier
The quality of the power amplifier depends to a large extent on its driving performance. This drive performance can easily be misunderstood as the output power, internal resistance and other technical indicators of the amplifier. We have no shortage of examples to see which tube amplifiers with only a few watts of power output can still play melodious music. However, their internal resistance is not very small, nor do they have good technical indicators. How much power the power amplifier can output indicates how much power reserve it has. This is not necessarily related to drive performance. The reason is as if the driving performance of a car is not necessarily related to whether the car has strong power. Unless this reserve power is not enough for the speaker vibration system to produce the required motion. The power amplifier has good driving performance, so that the speaker tray can accurately follow the signal waveform to move.

If we look at the speaker from another angle, we can think of it as a linear motion motor. In this way, we can further understand the importance of drive performance. Drive performance is related to the effectiveness of control. In many cases, the speaker's vibration system loses control, resulting in machine-like inertial vibration, which is considered to be the original low-frequency component of music. In fact, this is an example of poor driver performance. This type of sound is characterized by a single frequency as the main component, and maintains the same sound pressure amplitude in a short period of time. It is similar to acoustic resonance in a room, and we can easily recognize it from the sound of music.

The drive performance is not necessarily related to the device type. But in this respect to achieve the same performance, different device types still have some differences. The tube amplifier has an output transformer, and its driving force is relatively weak due to obstruction. If the amplifier has a large internal resistance, when matched with a high-compliance speaker, there will be a longer period of under-driving. In addition, because the amplifier uses an output transformer, the change in speaker impedance will be equivalent to the power amplifier tube having a load that alternately changes between capacitive and inductive. This easily leads to phase distortion of the signal. Faced with these situations, the drive design of the tube power amplifier will encounter more difficulties. If vacuum tube amplifiers are required to have the same driving performance, their design also needs to solve their relatively more unfavorable factors. Any power amplifier with an output transformer must encounter the above-mentioned problems. Avoid using high-compliance speakers and choose speakers with relatively flat impedance characteristics, which can reduce their requirements for amplifier drive performance. This choice is equally valid for power amplifiers of any type of device.

Drive performance is a technical topic. It refers to the ability of the amplifier to respond appropriately to the state to maintain its drive. This involves many aspects of control technology issues, which are not part of the content of this page and will not be discussed here.

The advantage of the high-power output amplifier is that after matching with the speaker, the output of the amplifier can have a wider range to maintain excellent electrical/acoustic conversion linearity. However, this only makes sense when comparing products of the same brand and the same series. Different brands or different series of products may have different circuit structures or use different technologies. Its performance has been different, and there is no such inevitability.

There is no absolute advantage in high-power output amplifiers. In the case of the same power output, the high-power amplifier will cause greater disturbance to the energy supply and cause greater interference to other devices. Adding power supply filter capacitors or modifying its internal power supply cannot effectively improve this interference. It needs to reduce the internal resistance of the power supply network to solve this problem. Find out how far the street grid transformer is from home, how many users the power supply line passes through, etc. These are all factors of the internal resistance of the grid and external interference. Certain power filters may also increase the internal resistance of the power supply network. In the case of the same internal resistance of the power grid, the interference of the full class A power amplifier to the power supply will be relatively small. Higher efficiency circuits will produce greater interference.

Class A power amplifier
There are two types of Class A power amplifiers, one is half A type, and the other is full A type. Most half-A amplifiers are high-power amplifiers. These semi-A amplifiers work in the Class A state when outputting 2~3W of power (some may be larger). If they exceed this range, they will switch to the Class B state. The power tube of the full A power amplifier works in the A state all the way. The single-ended drive tube amplifier is a full A type.

Compared with the class AB power amplifier, the power tube of the class A power amplifier still needs to perform voltage swing under the high current state when the output is weak. Therefore, the speed of state changes will be relatively low (longer response time). This circuit has a strong micro-motion filter, and the sound is relatively smooth. The intensity of the micro-dynamic filtering of the semi-Class A power amplifier will increase with the increase of the depth of the Class A state. When it reaches a certain level, the original details of the music will gradually decrease. Full Class A power amplifiers have stronger micro-dynamic filtering.

Due to the slower speed, the intermodulation distortion of the Class A power amplifier is larger. A small amount of intermodulation distortion products can change the timbre of the sound. As the number of frequency components of a music signal increases, the influence of intermodulation distortion increases. When playing some music with a wide frequency range (such as music played by a large orchestra), the sound will be messy. The degree of confusion is related to the overall phase characteristics of the audio system.

Cross Distortion
One of the main features of the Class A power amplifier is that the power device will not be turned off during the entire working process. This feature is easy to misunderstand, thinking that there is no cross distortion in Class A power amplifiers. On the surface, the power device is not off, and it seems that the cross distortion can be eliminated. However, this is not the case. The cross mentioned here includes two aspects. One refers to the transition of the on-off state of the power device. The other is the process in which the sine wave trajectory passes through the positive and negative alternating points of zero amplitude. The cross problem caused by the turning off of the power device has been solved in the AB circuit. Another sine wave cross distortion is a problem that can never be completely solved, especially for Class A amplifiers.

webassets/Sinewave--E.jpgWhy is sine wave crossover distortion a problem that cannot be completely solved? This is a basic topic. We need to explain from the formation of sine waves. In the Cartesian coordinate system, the vector whose starting point falls on the coordinate origin (0, 0), takes the starting point as a fixed point, and performs circular motion at a constant angular velocity. In motion, the magnitude of the end point of the vector on the Y axis (the horizontal projection of the end point of the vector on the Y axis) is the amplitude of the sine wave corresponding to a certain radian. At the zero crossing point (Y equals 0) of the sine wave, the circle tangent is orthogonal to the X axis. The circle tangent is the slew rate of the sine wave trace. The tangent of the circle at the zero crossing point is a vertical line, which means: the sine wave has an infinite slew rate at the zero crossing point. Regardless of the frequency of the sine wave, the situation where the slew rate is equal to infinity at the zero-crossing point will not change.

For technical reasons, there is currently no amplifier with a conversion rate equal to infinity, and such amplifiers will not appear in the future.

Under current technical conditions, the main problem of sine wave cross distortion is speed. On the contrary, when the output is weak, the state of the Class A power amplifier changes slowly, which is not conducive to improving the cross distortion of the sine wave. Therefore, the full class A power amplifier only has the characteristics of not turning off the power device. Not only can it not solve the problem of sine wave zero-crossing distortion, but it will also produce greater sine wave zero-crossing distortion under the same conditions. In addition, the non-turn-off feature allows the power device to withstand greater voltage swings. For existing power devices, this will have relatively poor linearity. In various test spectrograms, we can also see that the distortion of the Class A power amplifier is greater than the distortion of the Class AB power amplifier.

Above, we have made some explanations on the formation mechanism of sine wave crossover distortion. The technical indicators of sine wave distortion are usually measured on audio equipment. However, the audio equipment handles music electrical signals in actual work. Therefore, if we want to accurately assess the substantial impact of various distortions of audio equipment on music playback, we need to discuss them in conjunction with the characteristics of music electrical signals.

Tube amplifier
The biggest feature of the tube amplifier is that people have the fun of "playing", and the process is very convenient, and there is no requirement for technical knowledge in operation. If we do not have very high requirements for music details and use a tube amplifier, we can easily replace the tube to meet everyone's own sound preferences.

Regardless of whether the brand or model of the tube is the same, each tube has its own sound personality. However, the use of tubes to "play" sounds may only be suitable for preamplifiers. Because we rarely see the life of the power tube can exceed 2000 hours. After this time limit, its puberty will disappear, and the sound will change with the electrical characteristics of the tube. This is one of the biggest weaknesses of tube amplifiers. However, the Achilles heel of the tube amplifier does not lie in this.

Such problems can be found in audio systems with high-resolution capabilities. When the amplifier is replaced with a tube, you may Hard to get back the sound effect you originally liked. The electron tube used in the preamplifier can have a relatively long life cycle, and some brands can reach 10,000 hours. However, they will gradually age without being noticed, and at the same time, their voices will change accordingly.

Compared with other equipment, the speaker is the only component in the audio system that does not reach the ideal level of technical indicators. At the same time, it is also the most personalized component with the largest relative difference in electrical characteristics.

Impedance Characteristic. In the effective working frequency range, no loudspeaker can have a horizontally flat impedance characteristic. After the speaker is connected to a power amplifier with an output transformer, the change in speaker impedance will cause micro-dynamic distortion of the amplifier. If the audio system uses a power amplifier with an output transformer, choosing a speaker with a relatively flat impedance characteristic curve will help reduce this distortion.

Phase Characteristics. In the effective working frequency range, no loudspeaker can have horizontally flat phase characteristics. By adjusting the placement of the speakers, the phase characteristics can be corrected to a certain extent. A speaker with flat phase characteristics is more conducive to reproducing real sound, and its placement location can also have more choices.

Frequency Response Characteristics. In the effective working frequency range, no loudspeaker can have a horizontally flat frequency response characteristic. Frequency response characteristics will affect the timbre of the sound, but this is different from the change in timbre caused by distortion. This effect will be linear. It is only equivalent to the personality difference of the same instrument but different timbre, and basically does not cause the change of the sound state.

Speaker's edge
Compared with other side rings, rubber ring speakers have greater machine inertia. Changes in the state of the vibration system (paper tray, voice coil, etc.) will inevitably have a relatively long response time. This feature may have the following two situations when the amplifier drives the speaker. 1) Under-drive, there is a strong micro-dynamic filter, which will increase the ambiguity of the sound; 2) Over-drive, the vibration system will have overshoot. The chances of the above two situations in the speaker will increase as the inertia of the machine increases.

Compared with the rubber ring, the foam and cloth ring speakers have a smaller machine inertia, it is easier to obtain the cooperation of the power amplifier, and the probability of underdrive or overload is relatively low.

Whether the loudspeaker will appear under-driving and over-driving, depends on the interaction between it and the amplifier. As the inertia of the vibration system increases, the speaker's requirements for amplifier drive performance also increase.

Every time you use the speaker, you need to warm it up for a period of time before it can enter the linear working state. The loudspeaker has greater inertia and requires longer Warm-up time.

In the audio system, the cable is the most special component . A cable is an electronic circuit mainly composed of distributed components. The parameter values of the distributed component are uncertain and the range of dispersion is large. This has the characteristic that different wires will have different transmission characteristics. This can also be used to change the sound state and timbre. With the simple method of changing cables, you can get the kind of sound and tone you like relatively easily.

Each brand of interconnection cable has its own different sound characteristics. However, there may still be some differences in the sound effects of interconnecting cables of the same brand or even the same model but in different production batches. Signal interconnection lines basically have such characteristics. The same pair of interconnection lines are used in different systems, and the sound characteristics of the interconnection itself remain unchanged.

Balanced or Single Ended
Balanced signal transmission has stronger anti-interference ability and better isolation. Therefore, for signal transmission, the balanced mode always has better transmission quality than the single-ended mode. However, when choosing a signal port, we also need to consider other factors. If the device does not process the signal in a fully balanced manner, its input or output port needs to have a single-ended/balanced conversion or reverse conversion circuit. The quality of these circuits will affect the sound quality.

Generally, as long as one of the two equipments processes the signal in a fully balanced manner, the effect of balanced transmission will be better. If the two equipments are single-ended processing signals, you need to consider factors such as the transmission distance of the signal, the functional quality of the conversion circuit, and the isolation design of the equipment. Therefore, the signal terminal selection in this case often needs to be compared on a specific equipment to distinguish the pros and cons of the signal connection.

Speaker Cable
Each speaker has its own electrical characteristics, and there are significant differences between each model. Therefore, the speaker becomes the most personalized unit of the audio system. Unlike signal interconnection cables, speaker cables are a type of power transmission cable. Two issues, standing wave and real-time spectrum, will affect the sound state at the same time.

Musical electrical signals have a wide frequency band (10 octaves). The power transmission of each frequency component has its own standing wave. And the respective standing wave amplitude will vary according to the electrical characteristics of each speaker. Combined with the real-time problem of the frequency spectrum, after the music electric signal is transmitted, the waveform changes of different speakers will be different, and the sound state will also be different. In this way, the speaker cable will inevitably become a special component that matches the personality of the speaker.

For a specific speaker, using different cables, the sound will have different sound states. With this function, you can get your favorite sound effect relatively easily by changing the speaker cable.

Power Cord
To a certain extent, the power cord can be equivalent to a filter. From the power supply connection point to the connection line before the power transformer generates the magnetic field, there is always a certain degree of filtering. Different power cords may have different filtering characteristics. On the other hand, each equipment may also need different filtering characteristics to get the best results. Therefore, if the power cord of one equipment is used on another equipment, completely different results may occur.

Optical fiber can withstand electromagnetic interference, so it has certain advantages for long-distance signal transmission. And this advantage will increase with distance. However, the photoelectric conversion and reverse conversion circuits of the equipment will inevitably increase the jitter of the data stream. Therefore, whether the advantages of optical fiber can compensate for the increased jitter in the device data stream may vary from system to system. Optical fiber is more suitable for professional equipment for long-distance signal transmission over tens of feet. Household equipment only transmits signals over short distances. The advantages of optical fiber may be difficult to compensate for the defects caused by photoelectric conversion. Except in environments with strong electromagnetic interference, or the system must use a large number of digital (or high-frequency) cables, home music playback systems try to avoid using optical fibers to transmit digital signals.

LP turntable
The sound of LP originates from the physical structure. No matter how the sound quality is, the change of its sound pressure value will always have a transition time related to the physical structure. Therefore, LP can basically have the characteristics of soft sound. To maintain this characteristic, accurate stylus pressure is necessary.

To maintain accurate stylus pressure under dynamic (working) conditions, the strut mechanism of the vocal arm will play an important role. The arm strut mechanisms of different axial types have different following characteristics for the oscillating movement of the stylus, which specifically affects the timbre and sound state.

The stylus pressure will change with environmental factors, such as temperature, humidity, air pressure, etc. The pressure scale on the tonearm can only be used as a reference, and it is difficult to rely on it to obtain accurate stylus pressure. For users of LP turntables, there may be an unexpected gain with an accurate stylus pressure gauge.

RIAA Equalization (preset equalization) only has an agreement on the amplitude-frequency characteristics, but does not establish a specification for the phase characteristics. Different types of analog inverse equalization circuits have different phase characteristics. Therefore, different phono amplifiers may also have different phase characteristics. The phase characteristics will affect the sound state, however, this parameter usually does not appear on the product's technical indicators.

MC Transformer
For the MC step-up transformer, the input impedance of the MM amplifier is the load of the transformer. The capacitance on the load will have a considerable negative impact on the signal transmitted by the transformer. Under certain conditions, the transformer has the characteristics of capacitance/inductance conversion. This characteristic makes the capacitance on the load equivalent to an inductance in series with the transformer input. This will inevitably produce strong micro-dynamic filtering, causing serious loss of high-frequency components of the music signal.

Therefore, if the MC step-up transformer is used, you need to: 1) Minimize the input capacitance of the MM phono amplifier; 2) The output terminal interconnection line must be of low capacitance and shorten the length as much as possible. Regarding the first point, compared with semiconductors, electronic tube equipment generally has a larger input capacitance, so it is not easy to meet this requirement.

Audio CD
In the history of audio development, audio CD is the first music signal source that does not require the use of preset equalization in its standard. This provides a solid foundation for the signal source to obtain flat phase characteristics.

CD Disk
The sound of CD is also related to the physical structure, which is a bit similar to LP. However, the impact of CD structure quality on sound quality is quite different. In the extrusion process of CD disc production, the physical properties of the plastic inevitably lead to the slope of the pit edge with a short cooling time. This slope becomes the gray area between digital states 0 and 1. The length of the cooling time will change the width of the gray area, and specifically control the degree of data jitter. The data will have errors due to this jitter, and even after the data is decoded, there will be analog signals that do not match the original record. For information on the mechanism that causes CD data errors, see CD Disk Data Jitter.

CD optical disc media has such characteristics, which puts forward more stringent requirements on CD players. Even so, if some high-quality CD discs are paired with a high-quality CD player, the sound quality will still be better than other audio sources. For example, XRCD has good sound effects on many HI-End players.

The codes AAD, ADD, DDD marked on the CD disc not only indicate which stage of the technology is used in the music recording on the CD disc. Through these codes, we can also know whether the music signal on the CD is processed by an analog mixer. Analog mixer will affect the sound quality of music playback from the aspect of phase.

The sound quality of an audio CD will not only be affected by burning and post-production, but the structural quality of the disc will also affect the accuracy of the data and ultimately affect the sound quality. The production process of some CD discs shortens the cooling time to increase productivity. As a result, the quality of the pit structure representing the data is reduced. The amount of jitter of CD disc data has also increased. We can distinguish these CD discs by simply observing the transmittance and uniformity of the reflective layer. CD discs with relatively thick and uniform reflective layers usually have a longer cooling time. They will always have better structural quality.

D/A Converter
The setting time of multi-bit digital/analog chips is relatively short, and can be converted more accurately and faster, so the sound detail density is higher. Many early CD players were equipped with this type of chip.

The one-digit digital-to-analog chip has poor conversion accuracy and softer sound. One and multiple digital/analog hybrid chips are also used in recent CD players. The electrical characteristics of this type of chip are between one-bit and multi-bit chips, and are closer to multi-bit chips. CD players produced in later stages mostly use the above two types of chips.

In the post-production of CD Transports, they used a more mature digital signal processor (DSP). These DSP chips include a 32Kbit CIRC decoder memory, which can process data faster, so there are more time resources to correct more data errors. The decoder with 8x digital filter and multi-bit digital/analog chip is used as an analog unit with post-production CD transmission. This method combined with a CD player can usually achieve higher detail density.

External Clock
If the CD player uses an external clock, it needs to be equipped with a high-quality digital cable to transmit the clock signal in order to avoid the clock's large jitter after long-distance transmission (relative to the internal clock). We can use the stage depth in music playback to compare the jitter of the clock. A CD player with less clock jitter has a deeper stage and a quieter music background when playing music. At the same time, the environmental sound will be clearer and closer to the reality (using the environmental sound to identify the degree of restoration of the original music details, the easiest to get an accurate judgment. You can also use this method to compare the degree of jittering CD disc data).

For the Up-sampling, please refer to The quality of audio equipment.

SACD is another digital format music source that uses PWM (Pulse Width Modulation) encoding. At present, SACD players have not solved the problem of noise in the frequency band above 40KHz after DSD decoding. For audio systems using such signal sources, avoiding the use of amplifiers with broadband response characteristics can reduce the interference of such noise.

Compared with CD, SACD disc has a higher data storage density. Due to the use of plastic as the data carrier, data stream jitter is more difficult to control. Therefore, under the same technical conditions, SACD playback will be more prone to data errors. For 24bit/96KHz and DVD Audio digital music formats, because they also have a higher data storage density, they will also encounter this problem.

About various coding formats
For digital music formats such as CD, SACD, DVD Audio, 24bit/96kHz, many people have compared their sound quality. Regardless of whether the evaluation of these comparisons is positive or negative, their statement is basically correct. This includes companies that develop coding formats, and their claims about the superiority of coding formats are also correct. These conclusions and statements may contradict each other. The reason is that there are some differences between ideals and reality. In principle, 24bit/96kHz is better than 16bit/44.1KHz, which is inevitable. But need to meet such a condition: the playback device can read the correct data, and can accurately convert the data into an analog signal. In actual use, the extent to which the playback device satisfies this condition determines the sound quality and the size of the difference, which specifically affects the result of each comparison.

Another topic of the same nature is the difference in sound quality between CD and LP. After the CD appeared, the discussion on this issue did not stop. Until today, thirty years later, LP is once again sought after by fans. This phenomenon only reflects the actual CD in use, and there is indeed a gap between sound quality and LP that cannot be underestimated. Just like the accuracy of emotional expression that is essential in music playback. Of course, if we regard music as language, the accuracy of its emotional expression is also very important. We also compared the sound of CD and LP. In our comparison, audio CD music playback has better performance. This is reflected in some singing performances. We can feel that singers have appropriate emotional expressions when performing works, and music can more accurately depict the artistic conception of lyrics. This is because CD has a technical foundation that provides more musical details. When the player gets the correct data, the sound quality of an audio CD is of course better than that of an LP record. However, this result cannot represent a general phenomenon. This is also because the quality of CD and LP sound quality and the degree of difference are also restricted by the conditions of whether the playback device can obtain the correct data. So our comparison result may be different from others.

Data correctness
Many factors affect the gap between ideal and reality. With the consumer-grade playback equipment we currently use, and the use of plastic discs as data storage media, it is difficult for the player to correctly obtain the data on the CD with the data storage density of the CD. For digital music formats with higher data storage density, more errors will occur when the player reads the data. The superiority of these more advanced formats is not enough to make up for the loss of sound quality caused by data errors.

As more and more data is generated, players need to have a higher speed to process the data. As the operating speed of the device increases, the frequency of the bit clock must be increased accordingly, that is, the cycle time of the bit clock will be shortened accordingly. Under the same technical conditions, as the period of the bit clock is shortened, the jitter of the bit clock relative to the period time will increase accordingly. At the same time, as the bit clock period is shortened, the jitter of the data stream relative to the bit clock period will increase accordingly. In other words, as the working speed of the equipment increases, the probability of data errors also increases.

Analog signal accuracy
D/A chips produced by the same technology. Compared with low frequency, high frequency sampling will increase the error component of the analog signal after digital-to-analog conversion. The process of converting each digital data into an analog signal requires a short settling time (settling time) to make the analog value accurate. The voltage (or current) value after this stabilization time is the value that the analog quantity should have.

High-frequency sampling will have a greater number of settling times, and the amount of error (the amount within the settling time period) will occupy more time space. Therefore, when the sampling frequency is increased to a certain extent, the time for the accurate value of the analog quantity is shortened, and the time for the error value is increased. That is, the proportion of analog signals containing errors increases. The influence of the settling time on the accuracy of the analog signal will subsequently reach a level that cannot be ignored.

The above-mentioned problems are just some of the many shortcomings. They affect the sound quality of music playback in terms of data correctness. Although in theory, many digital music formats and sound quality enhancement modes have some advantages. However, due to various restrictions, they may not currently bring the expected results. There are some differences between ideals and reality, and it is necessary for our technicians to reduce these differences. CD disc data low-jitter processing Tai Chi backup is an improvement measure for audio CD data prone to errors in actual use. It is further confirmed that whether the playback device can obtain the correct data is the key to whether the advanced encoding format can exert its advantages.

Created by:Chen
Date: Sep-2014
Last revision date: Aug-2021

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