Ⅰ. Design principles of audio products
1. Design aesthetics
Design aesthetics is one of the important principles in the design and development of audio products. Good design can provide users with an excellent visual experience and differentiate products to help sales and brand building.
2. Degree of freedom
Audio products should have enough freedom to adapt to different environments and usage needs. For example, earphones must be able to adjust the appropriate volume to adapt to the use in different noise environments.
Ergonomics refers to the integration of product design with ease of use and the user's anatomical dimensions, mechanics, and physiology. Audio products like headphones must be ergonomic to ensure comfort and fit.
4. Sound quality
Good sound quality is one of the most important design principles of audio products. Designers should ensure that the sound of audio products can bring users an excellent listening experience.
Ⅱ. Common test indicators of audio equipment
Common test indicators for audio equipment mainly include dynamic range, frequency response, signal-to-noise ratio, total harmonic distortion plus noise, phase, crosstalk and level and other parameters.
1. Dynamic Range (DR)
Dynamic range refers to the logarithmic value of the ratio of the maximum undistorted output power to the static system noise output power when the audio system is replayed. It also refers to the relative ratio between the brightest and darkest parts of the output image of a multimedia hard disk player. Generally, the dynamic range of audio equipment with better performance is above 100dB.
Dynamic range is the ratio of the largest signal to the smallest signal that a device can handle. This concept is easily confused with the concept of "signal-to-noise ratio". It can be understood that the signal smaller than the noise amplitude cannot be restored correctly, but some equipment can cut off the noise and the small signal from some links when there is no signal or the signal is particularly low, so as to obtain a better signal-to-noise ratio index ( This is the basic principle of "Dynamic Noise Reduction"). At this time, it is still impossible to deal with small signals correctly, and the measurement of dynamic range can avoid such artificial optimization. The measurement of dynamic range is to use a small signal (generally a sine wave of -60dB/1000Hz) to output to the device, then filter out the signal, measure the noise and harmonic levels of the remaining frequencies, and compare it with the maximum signal, the result is Dynamic Range. It can be expected that the dynamic range will generally be lower than the signal-to-noise ratio. However, in the absence of special circuits or software to deal with noise, generally there is not much difference between the two, and they can be referred to each other.
2. Frequency Response (FR)
The frequency response measurement observes the output level generated after the level of different frequencies is input to the device under test. It is an evaluation standard for the frequency response capability of the digital-to-analog/analog-to-digital converter in audio equipment. It is usually input to the device with a sine wave of equal amplitude swept from very low frequency to very high frequency. If the response of the equipment is very flat, then the reflection on the frequency response curve should be that the output level of all frequencies is equal, the trajectory line has almost no change and the slope is close to zero. The simplest full-band response measurement can only select the extremely low and extremely high intermediate frequencies in the frequency band to be tested for testing. If the input levels at these frequencies are the same, the output level of the device under test is representative of its actual response to these frequencies.
In the low frequency and high frequency parts, the reconstruction of the signal is more difficult, so there is usually attenuation in these two frequency bands. The better the output quality of the device, the flatter the frequency response curve will be. On the contrary, this not only attenuates quickly at high and low frequencies, but also may appear jittering in general frequency bands.
3. Signal-to-noise ratio (SNR)
The size of the noise level often depends on how big your signal is. The SNR is a concrete reflection of the performance of this device. The input signal is usually the standard operating level of the device or the maximum undistorted output level. The signal-to-noise ratio results measured using the maximum undistorted output level are also called dynamic range because they describe the two extreme performance values of the device under test. Dynamic range has a somewhat different meaning for digital equipment, which is usually expressed in negative decibel values.
In the traditional SNR measurement method, we need to perform two measurements and a little calculation. First, we control the AP (or the gain of the DUT), find the level that makes the THD+N of the DUT reach the 1% distortion point as the reference level, and set the reference level to the AP by pressing the key F4, then turn off the generator, and read the The SNR can be read directly by setting the unit to dBr. When testing SNR, we must pay special attention to limiting the measurement bandwidth with filters.
4. Total Harmonic Distortion plus Noise (THD+N)
Harmonic distortion is the addition of some additional frequency signals in the audio signal. Harmonic frequencies are integer multiples of the original signal. Total Harmonic Distortion is the sum of all measurements of the harmonics of the device under test. Before FFT, it was difficult to measure its own THD without adding noise. After adding noise it becomes relatively easy and maneuverable. In addition, the value of THD+N is convenient and objective, so it is widely accepted by the public.
THD+N will vary according to the measurement bandwidth, so we need to use high-pass and low-pass filters to limit the measurement bandwidth, and indicate the actual bandwidth used in the test in the part of the result. We typically use bandwidths ranging from 20Hz to 20kHz. THD+N also varies with the level and frequency of the applied signal. So we typically measure equipment with an IF signal around 1kHz and a standard operating or output level.
In the audio industry, phase measurement refers to measuring the time offset of a periodic waveform (such as a sine wave) within a cycle based on a reference waveform. The reference waveform is usually selected from the same signal at different nodes within the system or related signals from different channels. Device input/output phase and channel-to-channel phase are the two most common phase measurements. The phase shift varies with frequency, so we usually plot the phase response at multiple frequencies or in a frequency sweep.
Usually, the phase difference is not sensitive to the level, so we can set the output voltage of the DUT to be higher than the noise floor without distortion. The phase difference between channels will change with frequency, so in order to fully reflect the phase difference information, we usually perform frequency sweep measurement.
In a multi-channel audio system, it usually occurs that the signal of one channel leaks into another channel in the form of low level. This cross-channel leakage signal is called crosstalk. It is usually expressed as the ratio between the leaked signal and the original signal. Crosstalk is very difficult to eliminate in actual devices.
Crosstalk is primarily the result of capacitive coupling between device channel conductors and typically exhibits an increase in frequency characteristic. Crosstalk results are usually just a single number. However, performing a frequency sweep test on a device can objectively reflect its actual crosstalk performance within the working bandwidth.
There are mainly the following types of test levels commonly used in audio equipment testing.
(1) Given output level, such as 1V, 1W or unity gain;
(2) It can produce fixed distortion level, such as 1% THD+N;
(3) Test the input or output levels specified in the documentation. When testing, we should choose the appropriate level to measure the equipment according to the different situations. So we must first be very clear about which level we should use.
Ⅲ. Audio evaluation
1. Audio evaluation method
There are two methods of evaluating the quality of reproduced sound, subjective evaluation and objective evaluation. The so-called subjective evaluation refers to the subjective evaluation of various sound effects by the listener. The objective evaluation refers to the use of instruments to test technical indicators.
However, due to the complex properties of musical sound quality, the subjective evaluation is relatively personal, and the existing audio testing technology can only reflect its fidelity from certain aspects. Therefore, so far there is no internationally recognized evaluation standard that can truly quantitatively reflect the fidelity of musical sound quality. But it is also reported that the International Telecommunication Union (ITU-T) has recently introduced a method for objectively evaluating sound quality. It is called the new measurement method of electronic ear. It can be used to objectively evaluate the sound quality of any audio equipment, and can also be used to detect defects in telephone communication speech coding systems.
2. Subjective evaluation of audio
Usually, we subjectively evaluate the various attributes of sound quality based on the changes and combinations of the three elements of musical sound quality, namely loudness, pitch, and pleasantness. For example, the sound is full when the low frequency is loud, the sound is bright if the high frequency is loud, and the sound is smooth when the low frequency is weak , the high frequency is weak and the sound is clear.
The bass is calm and powerful, thick and not muddy. The treble volume is moderate, there is a certain brightness, the reverberation is appropriate, and the distortion is small.
(2) Layered sense
The frequency response of high, medium and low frequencies is balanced. The high-pitched harmonics are rich, clear and slender without harshness. The midrange is bright and prominent, full without harshness. Bass is thick without twang.
(3) Sense of space
Although the sound of primary reflection and reverberation of multiple reflections lags behind the direct sound and has little effect on the sense of sound direction, the reflected sound always reaches the two ears from all directions, which has an important impact on the auditory judgment of the size of the surrounding space, making the human ear surrounded. This is the sense of space. A sense of space is more important than a sense of orientation.
It is mainly composed of the sense of surround, direction, and thickness of the sound. All kinds of sound fields in nature are full of three-dimensionality. It is one of the most important characteristics of the sound image of the simulated sound source.
The De Boer effect proves that the physiological characteristics of the human ear are: the human ear is on the symmetry axis of the two sound sources. When the sound pressure difference △p=0dB and the time difference △t= 0ms, the sound image of the two sound sources is the same, and the two sound sources cannot be distinguished. And when △p>15dB or △t>3ms, the human ear will feel that there are two sound sources, and the sound image will move to the sound source with higher sound pressure or the leading sound source, and every 5dB sound pressure difference is equivalent to the time difference of lms.
The Haas effect further proves that when Δ t=5ms~35ms, the human ear senses two sound sources. When the time difference △ t between the near reflected sound, lagging direct sound, or two sound sources is greater than 50ms, even if the loudness of the near reflected sound (also known as near or early reflected sound) or lagging sound is many times higher than that of the direct sound or leading sound, the direction of the sound source is still determined by the direct sound or leading sound.
According to the physiological characteristics of the human ear, as long as the intensity, delay, reverberation, and spatial effects of the sound are properly controlled and processed, there will be a certain time difference △t, phase difference △θ, and sound pressure in the two ears. Difference △P of the sound wave state, and make this state exactly the same as the sound wave state produced by the original sound source in both ears, people can truly and completely feel the three-dimensional sense of reproduced sound. Compared with monophonic sound, stereo sound usually has the characteristics of dispersed sound image, proper volume distribution of each part, high clarity, and low background noise.
(5) Sense of positioning
If the sound source is recorded and sent in different directions, such as left and right, up and down, and front and back, the received and replayed sound should be able to reproduce the direction of the sound source in the original sound field, which is the sense of positioning. According to the physiological characteristics of the human ear, the maximum time difference between direct sound from the same sound source and the two ears is 0.44ms~0.5ms, and there is also a certain sound pressure difference and phase difference. Physiological psychology has proven that bass frequencies between 20Hz and 200Hz are mainly located by the phase difference between the human ears; The 300Hz~4kHz midrange is mainly located by sound pressure difference; Higher treble is mainly located by time difference. It can be seen that the sense of location is mainly determined by the direct sound that first reaches both ears. The first reflected sound that lags to both ears and the reverberation sound that reflects multiple times from all directions mainly simulate the spatial surround feeling of the sound image.
3. Objective evaluation of sound quality
The objective evaluation of sound quality refers to the use of instruments to test technical indicators. Commonly used test systems are: CLIO, Klippel, Leap, etc. In the CLIO 10 test system, we can use MLS, SIN signals to test speaker delay curves, SPL, IMP, TS parameters, linear distortion, FFT analysis, etc. To know the performance of a loudspeaker, we need to test the parameters of SPL, IMP, and TS. Through these parameters, we can know the main performance of the speaker.
Ⅳ. Maintenance of audio products
Below are some suggestions for caring for your audio products.
1. Volume control: When using audio equipment, we should pay attention to properly adjust the volume to avoid damage to the equipment caused by too loud or too low volume.
2. Software update: We need to update the software and drivers of audio equipment in time to obtain the latest functions and performance optimization.
3. Cleaning and maintenance: We need to clean the audio equipment regularly to ensure that its surface is free of dust and dirt. We also use a soft damp cloth or a special cleaner to gently wipe the surface of the device. We want to avoid using too wet cloth or cleaning agent, so as not to cause damage to the equipment.
4. Equipment maintenance: We need to carry out equipment maintenance regularly, such as replacing the air filter, cleaning or replacing the audio network, etc., to ensure the best performance and reliability of the equipment.
5. Sun protection: We should avoid exposing audio products to direct sunlight to avoid damage to the equipment.
6. Moisture-proof: We need to keep the audio equipment dry and avoid placing it in a damp place. If the device is not used for a long time, we'd better put it in a dry place to avoid moisture problems.
7. Power connection: We need to ensure that the audio device is properly connected to the power supply and use a reliable power adapter or battery to avoid damage to the battery caused by overcharging or excessive discharge.