It has been more than 120 years since Edison invented the phonograph in 1877. From the mechanical recording / replay system that year to the current high-tech digital system, the progress of which is tremendous. However, the development of audio technology in the past 120 years is very uneven. In the 60 to 80 years after the invention of the phonograph, the development of audio technology has been quite slow, but it has also achieved certain results, such as recording and playback. The method replaced the mechanical method, and began to use multi-pole vacuum tubes and so on.

The rapid development of audio technology was in 927. Bell Labs of the United States announced the epoch-making negative feedback (negative feedback, NFB) technology, and audio amplifiers have entered a new era since then. The originator of the so-called High Fidelity amplifier should be traced back to the Williamson amplifier published in 1947. At that time, Mr. Willianson introduced a successful use of negative feedback technology in an article to design Hi Fi amplifiers to reduce distortion At 0.5% of the amplifier line, the sound of the timbre was unprecedented at the time, and it quickly swept the world, becoming an important milestone in the history of Hi Fi. Four years after the introduction of the Williamson amplifier, in 1951, the American Audio magazine published an article on "super linear amplifier". In June of the following year, another circuit design combining Williamson amplifier super linear amplifier was published. Because the ultra-linear design greatly reduces the nonlinear distortion, many people imitate it and form a boom again. The influence of ultra-linear design still exists today in the 21st century. It can be said that the Williamson amplifier and the ultra-linear amplifier mark the maturity of negative feedback technology in audio technology. Since that time, the design and types of amplifiers have been astounding. Technological advancements are unmatched in the first 70 years.

The specification of the amplifier is an important indicator to measure its performance. Of course, another important indicator is to receive goods by ear. Audiophiles often say that the specifications of audio equipment do not make much sense, but the sound of many amplifiers with good test data can't bear to listen. This is only half true. First of all, this excellent data is generally obtained when testing the prototype during the product development stage. Generally speaking, its performance will be discounted during mass production, depending on the grade of the equipment. The second is that although the current technology has greatly improved the performance of the amplifier, it is very difficult to amplify the audio signal of 20 ~ 20KHz that the human ear cannot detect distortion. Moreover, the so-called performance specifications of general amplifiers are only Given few data, most of them are only measured under certain conditions. Not enough to reflect the basic performance of the amplifier.

The methods used to evaluate the technical specifications of the amplifier are divided into dynamic and static. The static specifications refer to the indicators obtained by measuring the sine wave in the steady state. This is actually the frequency analysis method in Classical Control Theory. It has been in use since the 20th and 30th century. Test items include frequency response, harmonic distortion, signal-to-noise ratio, intermodulation distortion and damping coefficient. Dynamic specifications refer to the indicators measured with more complex signals such as square waves and narrow pulses, including phase distortion, transient response and transient intermodulation distortion. The dynamic test is actually similar to the transient response test common in industrial automatic control systems, except that the industrial test commonly uses a step signal (Step Signal) and the audio test uses a shortened step signal-square wave. To reflect the quality of the amplifier in general, dynamic testing and data must be comprehensively considered. As for the human ear audition, because it contains more subjective factors, I do not intend to discuss it in detail here. Since most manufacturers usually give only a few parameters to their products, I hope to take this opportunity to introduce some of the more important audio equipment specifications to facilitate newcomers and some non-engineering people. Sound technology has a deeper understanding.

Frequency response

Among the many technical indicators, frequency response is the most familiar specification. Part of the amplifier. In theory, it is sufficient to achieve a flat frequency response of 20 to 20,000 cycles, but the overtones (harmonics) contained in the real musical sound may exceed this range, plus to improve the performance of transient response, so The amplifier requires a higher frequency response range, such as from 10 Hz to 100 kHz. It is customary to stipulate the frequency response range: when the output level drops by 3 dB at a low frequency point, the point is the lower limit step rate, and also drops by 3 dB at a high frequency point, then it is set as the upper limit frequency . This decibel point has another name, called Half Power Point. Because when the power drops by half, the level just drops to understand the situation decibel. It must be pointed out that although the half power point has certain significance for some electronic devices and automatic control systems, it is not necessarily suitable for audio equipment, because the human ear can achieve a resolution of 0.1 dB. So there are some high-end equipments that are rated at 20 to 20K to reach plus or minus 0.1 decibels, which may actually be higher after the nominal 10 to 50K + 3DB specifications. Incidentally, the frequency response curve diagram actually has two, in the control project "Bode plot" (Bode Plot). The amplitude-frequency curve diagram is our common frequency response diagram, and the other is called the phase-frequency curve diagram, which is used to represent the degree of phase distortion (phase distortion) produced by different frequencies after passing through the amplifier. Phase distortion refers to the time difference (phase difference) generated by the signal from the amplifier input to the output. This time difference is naturally as small as possible, otherwise it will affect the work of the negative return line. In addition, the phase distortion is also related to the transient response, especially has a close relationship with the transient to modulation distortion that has been paid more and more attention in recent years. For Hi Fi amplifiers, the phase distortion should be at least within the range of 20 ~ 20KHz + -5%.

Harmonic distortion

After any natural physical system is disturbed by the outside world, there will be a periodic vibration that shows attenuation. For example, if a half-meter-long string at both ends is plucked in the middle, a vibration wave of 1 meter in wavelength, called a fundamental wave (Fundemental), will be generated. In addition to the large swing of the string along the center point In addition, the line itself makes many small vibrations that are difficult to detect with the naked eye, and its frequency is generally higher than the fundamental wave, and more than one frequency. Its size is determined by the physical characteristics of the string. In physics, these vibration waves are called harmonics. In order to facilitate distinction, harmonic waves generated by musical instruments are often referred to as overtones. In addition to the harmonics generated by the signal source, when the vibration wave propagates if it encounters an obstacle and produces reflection, diffraction and refraction will also generate harmonics.

Either the fundamental wave or the harmonic itself is a "pure" sine wave (Note: A sine wave is a periodic function composed of a positive half cycle and a negative half cycle, but its negative half cycle must never be called a negative sine wave!) But they When combined together, many hall-shaped waveforms are produced. Figure 3: It is a waveform composed of a fundamental wave and a second harmonic (the frequency is doubled and the amplitude is half). The square wave that everyone is familiar with is composed of a sine wave fundamental wave plus a large number of harmonics (singular), which also explains why the square wave is often used as a test signal.

The amplifier circuit is full of various electronic parts, wiring and solder joints. These things can more or less reduce the linear performance of the amplifier. When the music signal passes through the amplifier, the nonlinear characteristics will cause the music signal to be distorted to a certain degree. According to the aforementioned theory, this is equivalent to adding some harmonics to the signal, so the distortion of this signal deformation is called harmonic distortion. It is not difficult to understand why harmonic distortion is often expressed as a percentage. A small percentage means that the amplifier generates less harmonics, which means that the signal waveform is distorted to a lower degree. The components of harmonics generated by different physical systems are also different. But there is one thing in common, that is, the higher the harmonic frequency, the smaller the amplitude. Therefore, for audio amplifiers, it is the two to three harmonic distortion components whose frequency is closest to the fundamental wave that cause obvious audible distortion of the sound.

When calibrating the harmonic distortion of products, manufacturers usually only give one item of data, such as 0.1%. However, the harmonics generated by the amplifier are not a constant, but a function related to the signal frequency and output power. Figure 4 shows the relationship between the harmonic distortion of two typical transistor two-channel amplifiers and the signal frequency. Figure 5 shows the relationship between harmonic distortion and output power of a transistor amplifier with an output of 100W. It can be seen from the figure that when the output power is close to the maximum value, the harmonic distortion increases sharply. Because the transistor will clip when it is close to Overload. Cutting off the top of a signal flush is obviously a severe waveform distortion. Harmonic distortion will naturally increase significantly.

Harmonic distortion is not completely useless. One of the reasons why the sound of the amplifier is soft and beautiful is that the amplifier mainly produces even harmonic distortion. That is, the frequency is a harmonic of 2'4'6'8 '... fundamental frequency. Because the harmonic level is inversely proportional to the frequency, the second harmonic has a large amplitude and a large impact. The rest has a small amplitude, so the impact is large, and the other has a small amplitude, so the impact is slight. Although the second harmonic is technically speaking It is distortion, but because its frequency is double the fundamental wave, it is just an octave, that is to say, the right and the fundamental wave form a pure octave in music. We know that pure octave is the most harmonious and beautiful harmony. Therefore, it is not difficult to understand that the amplifier has a sweet sound and rich music. In the 1940s, many “smaller” radios deliberately added a considerable degree of second harmonic distortion. The purpose is to create "heavy bass" to please consumers. The right voice can be very enjoyable, but it is completely contrary to the requirements of high fidelity.

Signal to noise ratio

Signal to noise ratio (Signal Noise RaTIo), referred to as signal to noise ratio or signal to noise ratio, refers to the ratio of useful signal power to useless noise power. Normally, the power is a function of current and voltage, so the signal-to-noise ratio can also be calculated using the voltage value, that is, the ratio of the signal level to the noise level, but the calculation formula is slightly different. Calculate the signal-to-noise ratio with power north rate: S / N = 10 log Calculate the signal-to-noise ratio with voltage: S / N = 10 log Because the signal-to-noise ratio is logarithmic with the power or voltage, it is necessary to increase the signal-to-noise ratio Significantly increase the ratio of output value to noise value. For example, when the signal-to-noise ratio is 100dB, the output voltage is 10,000 times the noise voltage. For electronic circuits, this is not an easy task.

If an amplifier has a high signal-to-noise ratio, it means that the background is quiet. Due to the low noise level, many details of weak sounds covered by noise will appear, which will increase the floating sound, enhance the sense of air, and increase the dynamic range. There is no strict discriminant data for measuring whether the signal-to-noise ratio of the amplifier is good or bad. Generally speaking, it is better to be above about 85dB. Below this value, it is possible to hear noticeable in the music gap under certain high volume listening situations Noise. In addition to the signal-to-noise ratio, the concept of noise level can also be used to measure the noise level of the amplifier. This is actually a value of the signal-to-noise ratio calculated by voltage, but the denominator is a fixed number: 0.775V, and the numerator is Noise voltage, so the difference between noise level and signal-to-noise ratio is: the former is an absolute value, the latter is a relative number.

Behind the specification table data in many product manuals, there is often an A, meaning A-weight, that is, A weighting, weighting means that a certain value is modified according to certain rules and weights, due to human The ear is particularly sensitive to the intermediate frequency, so if the mid-band signal-to-noise ratio of an amplifier is large enough, even if the signal-to-noise ratio is slightly lower in the low and high frequency bands, the human ear is not easy to detect. It can be seen that if the weighting method is used to measure the signal-to-noise ratio, its value must be higher than that without the weighting method. In terms of A-weighting, its value will be higher than the non-weighting of the dating decibels.

Intermodulation distortion

As the name implies, Intermodulation Distortion (IntermodulaTIon DistorTIon) refers to the distortion caused by the intermodulation of signals. The term modulation originally refers to a technology used in communication technology to improve the efficiency of signal transmission. Since the original signal containing sound, image, text, etc. is "added" to the high-frequency signal, then the comrade sends the synthesized signal. This process and method of "adding" high and low frequencies is called a modulation technique, and the resulting signal is called a modulation signal. In addition to retaining the main characteristics of high-frequency signals, modulated signals also contain all the information of low-frequency signals. The process of generating intermodulation distortion is essentially a modulation process. Since an electronic circuit or an amplifier cannot achieve perfect ideal linearity, when signals of different frequencies enter the amplifier and are amplified at the same time, under the action of nonlinearity, Each signal of different frequency will be automatically added and subtracted, resulting in two additional signals that are not in the original signal. If the original signal has three different frequencies, there will be 6 additional signals, when the original signal is N , There will be N (N-1) output signals. It is conceivable that when the input signal is a complex multi-frequency signal, such as orchestral music, the number of extra signals generated by intermodulation distortion is amazing!

Since the intermodulation distortion signals are all harmonic frequencies of the music frequency, they are completely the same as the natural sound, so the human ear is sensitive to this. Unfortunately, in many amplifiers, the intermodulation distortion is often greater than the harmonic distortion. This is partly because harmonic distortion is generally easier to deal with.

Although intermodulation distortion and harmonic distortion are also caused by the nonlinearity of the amplifier, from a mathematical point of view, the two are also added some additional frequency components in the positive symbol, but they are actually not the same, simple It is said that harmonic distortion is the distortion of the original signal waveform. Even a single frequency signal will produce this phenomenon through the amplification line, but intermodulation distortion is the mutual interference and influence between different frequencies. Measuring intermodulation distortion is far It is more complicated than measuring harmonic distortion, and there is no unified standard yet.

The world premiere of the technical zone! ROHM has developed the power supply IC "BD372xx series" for high-quality audio. A practical guide for the purchase of home wireless routers. Understanding the circuit diagram and working principle of the audio. Talking about the "frequency response curve" in the audio. Deep dismantling report of the M0pro speaker: both internal and external

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Power amplifier parameter index (below)

Transient intermodulation distortion

Published on 2006-04-17 23:57 • 252 times read
Power amplifier parameter index (below)
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