How to choose a suitable audio device in ordinary multimedia and home projection is a headache for many engineering companies (weakly integrators). Today's projector market is getting thinner and thinner, and the price is getting lower and lower. As the price of the projector market is becoming more and more transparent, it is difficult for engineers to use the projector equipment to maintain the original profit point of the project. Integrators only have more brains in the control, audio, control cabinets, construction costs, etc., in order to improve the overall profit margin.
Therefore, audio is one of the factors that control costs, satisfy users, and increase profits. Price and appearance are the preferred targets of many businesses, and sound quality effects and features are often placed behind them. Therefore, in the later use, there will be many complaints from users, such as: unsound sound, large distortion, too much noise, vocals such as broken drums, music is not rich, no level, no REC (recording) port, no The OUTPUT (auxiliary output) port or the INPUT (auxiliary input) port is not enough, and so on. A high-priced, high-performance video device is affected by audio products, resulting in poor maintenance of late relationships and a gradual loss of customer resources. So choosing a suitable sound reinforcement device is important.
So how do you choose a sound that is more cost-effective and more suitable for users? (General multimedia such as: teaching, conference, demonstration hall, etc.)
1. It is not necessary to have a good brand of expensive audio, as long as it can meet the needs of general vocal amplification and music playback. The general multimedia we talked about here is the simplest one with video playback. The sound pressure level of the sound field is more than 75dB.
2, the indoor environment is about 1W per square meter (with the power of the speaker) is enough, which has a great relationship with the conditions of the sound, can not be determined. In fact, many of our daily orders have not met this requirement.
3, the appearance is beautiful and professional. This looks good and can be a good price. Usually, the karaoke appearance is the best. First, how many households have used it. It is also considered that the price is not low and can sell a good price. Many dealers and contractors believe that the purchase price is not cheap. That is because all the OK amplifiers on the market are designed for karaoke bars and clubs. There is no amplifier designed specifically for projection, but for projection. In the matching power amplifier, the variable frequency, digital effect, liquid crystal display, equalization, etc. are simply the extra cost increase, then the microphone input is changed to less, the auxiliary input is also less, the cover is thinner, the cost is greatly reduced, and the market price is 500-1000 can be reduced to more than 350 points, things are simple, as long as the electronic components are unchanged, the machine life is longer. In addition, the built-in preamplifier can directly insert the microphone and bring a simple effect.
4, the function is simple, easy to use. Most households do not understand the product, the adjustment knob (potentiometer) is more troublesome, but the more chaotic, the after-sales service is also more, in fact, just not used, but also always blame the equipment is not good. Therefore, many users say that the fool is the best, there are several buttons on the machine, such as the total volume of the MIC (microphone), MUSIC (music), the size of the effect.
5. The auxiliary power amplifier should be full. There are several terminals for music input, and now there are some occasions that like various audio source playback devices, computer, DVD, MP3 or TV. At the same time, many units make reports and engage in internal recordings, so REC (recording port) is also indispensable.
6, two channels, there are obvious high and low sound. Nowadays, many companies like to use public broadcasting equipment as sound reinforcement, and they feel that it is easy to match. In fact, I don't know that such constant voltage amplifiers and speakers are usually used for long-distance signal transmission. The distortion is quite powerful. The built-in constant voltage transformer does not affect the sound quality, and the cost of the transformer is not low, which naturally increases our purchase cost. And it's monophonic in itself, playing music without stereo and layering. Usually, DVDs, computers, etc. are left and right channels.
How to choose the matching audio http:// target=_blank>hifi/ target=_blank>HIFI.com/thread-37051-1-1.html
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Room equalizers are typically fine-tuned with a pink noise generator and real-time spectrum analyzer. The room equalizer is mainly used to correct and compensate the room frequency characteristics. Therefore, the consistency of the auditorium environment and the actual listening environment should be ensured during commissioning. In addition, the adjustment of the room equalizer sometimes needs to be combined with the adjustment of the speaker layout.
The room equalizer compensates for environmental frequency characteristics by changing the frequency characteristics of the signal. Changes in frequency characteristics inevitably lead to changes in phase characteristics, causing phase distortion. When the adjustment of the room balance is too large, especially in a certain wide band, the adjustment effect must be achieved with a large adjustment amount. Although the frequency characteristics of the room are corrected, the phase distortion is caused. Relationships, the sense of hearing will become very poor, this situation will be more prominent for stereo systems. In the case of poor sound conditions, room equalizer adjustments can sometimes only compromise between frequency characteristics and hearing. The flat result of the strong frequency characteristics is sometimes self-defeating. The best way is to improve the acoustic properties of the room itself.
1. Debugging process (1) Using pink noise as the system input test signal, this noise is obtained by white noise passing through a -6dB/oct filter. Compared with white noise, pink noise has a higher frequency of low frequency. Because the pink noise energy distribution is close to the real music signal, it is often used as a test signal for sound engineering and audio equipment. The power capacity of the speaker is also generally pink noise. Generally, the frequency response of the laser player above the mid-range can be achieved at 20Hz-20kHz+0.5dB, which can meet the test requirements.
(2) Input the pink noise into the mixer, adjust the mixer to the standard output level, usually OVU, the output level is +4dB. It should be noted that the EQ of the equalizer on the mixer is adjusted to be flat, that is, all At the zero position, the frequency of each segment of the test signal is neither boosted nor attenuated. The room frequency adjustment potentiometers of the room equalizer are also temporarily placed in the zero position. Slowly increase the power amplifier volume adjuster to hear the pink signal, and monitor it with a sound pressure meter until the sound pressure level of the powder noise signal in the hall reaches about 85dB.
(3) Place the measuring microphone in the center of the hall, and the selector switch on the spectrum analyzer is placed in the “OCT†file (this file is the octave filter file, which corresponds to the characteristics of pink noise). At this time, the LED display on the real-time spectrum analyzer is the frequency characteristic curve of the listening environment. The flatter it is, the better the frequency characteristics of the room building sound.
(4) Adjust the frequency boost/attenuator at each point on the equalizer so that the frequency characteristic curve on the spectrum analyzer is in a straight line.
After the above debugging is completed, it is generally necessary to "smooth" the equalization curve on the equalizer, which is mainly to prevent excessive phase distortion caused by the equalizer being adjusted to the sawtooth frequency characteristic.
2, room equalizer adjustment points (1) in the low frequency band of 20-50Hz and high frequency band above 14kHz, the frequency characteristics do not have to be strong, especially for the low frequency band. Because the general speaker is difficult to extend to 20Hz, it is good to be able to reach 40Hz. The flatness of the low-frequency characteristics and the improvement of the ultra-low frequency will make the speaker "out of control" due to the excessively extended low frequency, and the distortion will be intensified.
(2) The adjustment of the room equalizer should always consider the contradiction between the flatness of the frequency characteristics and the minimization of the phase distortion, and make a compromise.
(3) In the case where there are obvious "peaks" and "valleys" in the frequency characteristics of the acoustic environment, consideration should be given to changing the position of the speakers and trying to change the sound characteristics.
(4) The adjustment of the room equalizer is a very detailed work, which requires repeated adjustments to be finalized. This is because in the adjustment process, it is often necessary to make some adjustments to the speaker placement and sound construction environment, and the equalizer will be mutually restrained when adjusting.
Objectively speaking, the role of the room equalizer is limited, and the defects of the sound environment cannot be expected to be completely solved by the room equalizer. The smaller the balance, the better the sound quality will be. In the absence of a pink noise generator and a real-time spectrum analyzer, the audio signal generator can be used to send the same amplitude signal to the system at each frequency point on the selected room equalizer, and the sound pressure meter can be used to test The sound pressure in the field and the adjustment of the room equalizer enable the input signals at each frequency point to produce the same sound pressure level in the field. The actual effect of this type of debugging is worse than using standard pink noise. Therefore, professional units should be equipped with pink noise generators and real-time spectrum analyzers whenever possible. Adjustment of the room equalizer in the sound system http:// target=_blank>hifi/ target=_blank>HIFI.com/thread-35933-1-1.html
(Source: China hifi / target = _blank > HIFI audio network)
1. Subjective perception of the frequency domain The most important subjective sensation in the frequency domain is the pitch. Like the loudness, the pitch is also a subjective psychological quantity of hearing. It is the attribute of the auditory judgment of the sound level. The difference between the pitch in psychology and the scale in music is that the former is the tone of pure tone, while the latter is the tone of a composite sound such as music. The tone of the composite sound is not simply a frequency analysis, but also a function of the auditory nervous system, which is influenced by the listening experience and learning of the listener.
2. Subjective feeling of time domain China hifi/ target=_blank>hifi/ target=_blank>HIFI audio network provides the most comprehensive domestic and international audio brand information and information for audiophiles, HiFi audio, headphone trend, car audio, civilian audio modification Technical and professional audio product evaluation, etc. Provide the most authoritative audio brand information and technology exchange, learning platform!
If the length of the sound exceeds approximately 300 ms, then the increase or decrease in the length of the sound does not contribute to the threshold change of the hearing. The feeling of tone is also related to the length of time of the sound. When the sound lasts for a short time, you can't hear the tone, just hear the "squeaky" sound. The duration of the sound is lengthened to have a feeling of tone. Only when the sound lasts for more than tens of milliseconds, the perceived pitch can be stabilized, and another subjective characteristic of the time domain is the echo.
3. The subjective feeling of the spatial domain The human ear has obvious advantages over the binaural listening. It has high sensitivity, low listening valve, a sense of direction to the sound source, and strong anti-interference ability. In stereo conditions, the sense of space obtained by listening to speakers and listening to stereo headphones is different. The sound heard by the former seems to be in the surrounding environment, while the sound heard by the latter is located inside the head, in order to distinguish between the two. The sense of space, the former is called orientation, the latter is called positioning.
4. Weber's law of hearing Weber's law shows that the subjective sensation of human hearing is proportional to the logarithm of objective stimuli. When the sound is small and the amplitude of the sound wave is increased, the subjective feeling volume of the human ear is increased by a large amount; when the sound intensity is large and the same sound wave amplitude is increased, the subjective feeling volume of the human ear is increased less.
According to the above listening characteristics of the human ear, an exponential potentiometer is required as a volume controller when designing the volume control circuit, so that the volume is linearly increased when the potentiometer is rotated uniformly.
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5. Hearing Ohm's Law The famous scientist Ohm discovered the Ohm's law in electricity, and he also discovered the Ohm's law of human hearing. This law reveals that the hearing of the human ear is only related to the frequency and intensity of the sounds in the sound. Relevant, regardless of the phase between the partial sounds. According to this law, the control of the recording, playback, and the like in the sound system can not consider the phase relationship of each of the partial sounds in the complex sound.
The human ear is a frequency analyzer, which can separate the homophonic sounds in the polyphony. The sensitivity of the human ear to the frequency is very high. At this point, the human ear has higher resolution than the eye, and the human eye cannot see various kinds of white light. Colored light component.
6. Other sounds in the masking effect environment will cause the listener to reduce the hearing of a certain sound, which is called masking. When the intensity of one sound is much larger than the other, when the two sounds are both large and the two sounds exist at the same time, one can only hear the sound of the sound, but not the other. The amount of masking is related to the sound pressure of the masking sound, and the sound pressure level of the masking sound increases, and the masking amount increases. In addition, the masking range of the low frequency sound is larger than the masking range of the high frequency sound.
This auditory characteristic of the human ear provides an important inspiration for designing noise-reducing circuits. In tape playback, there is such a listening experience. When the music program is continuously changing and the sound is loud, we will not hear the noise floor of the tape, but when the music program ends (blank tape), we can feel it. The "å’..." noise to the tape exists.
In order to reduce the influence of noise on the sound of the program, the concept of signal-to-noise ratio (SN) is proposed, that is, the signal strength is required to be sufficiently larger than the noise intensity, so that the listening does not feel the presence of noise. Some noise reduction systems are designed using the principle of masking effects.
7. The basic principle of the binaural effect binaural effect is this: if the sound comes directly from the listener, at this time, because the distance from the sound source to the left and right ears is equal, the time difference (phase difference) between the sound wave reaching the left and right ears, The timbre is zero, and the sound is heard from the front of the listener instead of being biased to one side. When the sound is different, you can feel the distance between the sound source and the listener.
8. The Haas effect Haas test proves that when two sound sources are simultaneously sounded, the feeling of binaural listening is different according to the delay amount of one sound source and the other sound source, and can be divided into the following three types. The situation to explain:
(1) When the delay time of one sound source and the other sound source of the two sound sources is within 5~35mS, it is as if the two sound sources are combined into one, and the listener can only feel the sound source of the previous one. Existence and direction do not feel the existence of another sound source.
(2) If one sound source delays another sound source 30~50mS, the existence of two sound sources can be felt, but the direction is still determined by the predecessor.
(3) If the delay of one sound source is greater than 50mS of the other sound source, the simultaneous presence of the two sound sources can be felt, the direction is determined by each sound source, and the hysteresis sound is a clear echo.
The Haas effect is one of the foundations of stereo system orientation.
9. The Lloyd's effect Lloyd's effect is a psychoacoustic effect of the stereo range. The Lloyd's effect reveals that if the delayed signal is re-inverted and superimposed on the direct signal, it will produce a distinct sense of space. The sound seems to come from all directions, and the listener seems to be in the band.
10. The keyhole effect mono recording system uses a microphone to record, the signal is recorded on a track, and one amplifier and one speaker are used for playback, so the sound source is a point source, just like the listener passes the door. The keyhole on the upper part listens to the symphony in the room, which is the so-called keyhole effect.
11. The bathroom effect has a personal feeling when it comes to the bathroom. The sound in the bathroom, the reverberation time is too long and excessive, this phenomenon is called the bathroom effect in the sound quality description of electroacoustic technology. When the low and medium frequency segments are exaggerated, there is resonance, the frequency response is not flat, and the 300Hz boost is excessive, the bathroom effect will occur.
12. Doppler effect Doppler effect reveals the listening characteristics of moving sound: When there is relative motion between the sound source and the listener, it will feel that the sound determined by a certain frequency has its tone changed. When the source approaches the listener, it is a slightly higher frequency tone, and when the sound source is gone, it is a slightly lower frequency tone. The amount of change in this frequency is called the Doppler shift. The moving sound source is greater at the same distance from the listener than when it is not moving, and the removed sound source produces less intensity, usually the sound source is concentrated toward the moving direction.
13. The Li Kai test Li Kai test proves that when the phases of the two sound sources are opposite, the sound image can exceed the two sound sources and even jump behind the listening sound.
The Li Kai test also suggests that as long as the intensity and phase of the two sound sources (left and right channel speakers) are properly controlled, a wide range (angle, depth) of the sound image moving field can be obtained.
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