Natural reverberation control in music studio
- Categories:Industry News
- Time of issue:2018-04-12 11:38
Natural reverberation control in music studio
In the past two decades, the music recording studio, a dedicated place for recording music, has developed rapidly and its format is constantly changing. It cooperates and promotes each other with recording production technology and sound pickup technology, which greatly promotes the development and update of sound quality processing equipment and technology. During this period, natural reverberation music recording studios with adjustable reverberation time, short reverberation music recording studios, strong sound absorption music recording studios, and active end-silent end music recording studios appeared successively. At the same time, auxiliary facilities such as Booth and soundproof screens are widely used in the recording studio. Nevertheless, natural reverberation music recording studios still play an irreplaceable and important role in music recording.
1. The reverberation time of a large-scale music studio in my country
2. Room volume: The long reverberation natural reverberation music recording studio used to record serious music such as symphony requires a considerable volume. This is not only a requirement for reverberation time and sound diffusion, but more importantly, to avoid The sound in the room is saturated.
The so-called room sound saturation means that the room sound pressure level is too high. Excessively high sound level is the sound "explosive" and deafening in the sense of hearing; and for music with a wide frequency distribution (especially serious music such as symphony), it is possible to pass the highest microphone in some frequency bands. The allowable sound level, therefore, is difficult to correct by the sound attenuation after the microphone. It is known that the size of the indoor sound level mainly depends on the room constant and the sound power of the sound source. For a room with a smaller volume, if you want to ensure a longer reverberation time, it will inevitably reduce the sound absorption of the indoor surface. Under the condition of the same sound source sound power, the indoor sound pressure level will inevitably increase accordingly; roughly speaking, the highest sound level for listening to music is about 100 decibels. Common sense tells us that a piano with a sound power of about 0.4 watts is played in a normal room (reverberation time of about 0.8 seconds) and a large symphony orchestra (sound power of about 60 watts) is played in a concert hall with a reverberation time of 2 seconds. , The listener thinks that their sound levels are appropriate in the sense of hearing. If the performance environment of the two is swapped, the sound level of the former is too small (so many concert halls have to adopt sound reinforcement systems for it), and the sound level of the latter (if possible) will inevitably reach intolerable levels. The general view is that a music recording studio that fully utilizes natural reverberation actually works best in a concert hall with a volume of more than 10,000 cubic meters.
An effective way to solve the problem of sound saturation is to appropriately increase the sound absorption in the room. The increase in the sound absorption coefficient of the indoor boundary surface is equivalent to increasing the volume of the room in effect, but it also inevitably reduces the indoor reverberation time. As far as recording is concerned, it can of course be supplemented by artificial reverberation in practice, but such a recording studio can no longer be used as a natural reverberation type.
3. Diffusion of the room: Although it is difficult to achieve a diffuse sound field in the strict sense, most of the body shapes are irregular or the proportions of length, width, and height are appropriate. Large-scale music recording studios with properly arranged sound-absorbing or reflective surfaces in the room may satisfy Basic requirements for diffused sound field, such as no clear and intermittent reflected sound, sound field distribution is basically uniform, better directional diffusion (d value above 0.9), etc.
It is worth noting that, due to the size of this type of music recording studio, if it is not handled properly, it is likely to lack the necessary early reflections. Although the exact relationship between early reflections and sound quality is still unclear, it is certain that it will have an important impact on the feelings of intimacy, grandeur and intensity of music. Indoors, early reflections, direct sound, and reverberation also play an important role in the perception of distance and room size. Even if the reverberation time is appropriate, if the pickup point lacks the early reflections within 50 milliseconds, there may also be sound quality problems, such as "floating" sound.
4. Reverberation radius: Although the reverberation radius of a room is not an independent parameter describing the acoustic state of the room, it has very important practical significance for describing the reverberation conditions at different locations in the room. In other words, with the help of the reverberation radius, different amounts of reverberation can be picked up at different locations in the room until the maximum amount of reverberation determined by the room is reached.
When recording in a natural reverberation music studio, a basic requirement is to keep as much information as possible when the music is played. Since the volume of this type of recording studio is quite large, and the reverberation time is relatively long, the theoretical value of the reverberation radius and the measured value will not differ too much. This allows the reverberation radius value to be obtained from the volume and reverberation time of the recording room. Using the concept of reverberation radius and properly selecting the pickup point, it is possible to successfully pick up the sound of the entire band with only one microphone. The usual practice is to first base on the reverberation radius, and then make specific adjustments according to the sense of hearing to accurately select the distance between the microphone and the sound source and the specific location of the microphone.
It must be pointed out that for a large music studio with a certain reverberation time, the reverberation radius is not a fixed value in the entire frequency range of the band. It is not only related to the frequency characteristics of the reverberation time in the recording studio, but also related to the directivity of the musical instrument and the directivity of the microphone.
Without considering the directivity of the microphone (this will be discussed separately in the next chapter), only in terms of the sound source and room factors, because the reverberation time of the low frequency of the music studio is longer than that of the intermediate frequency (500 Hz) and high frequency . When the volume is constant, the corresponding reverberation radius will be shorter than the middle and high frequency; the radiation characteristics of the musical instrument will show quite obvious directivity due to the difference in frequency. The directivity of most musical instruments increases as the frequency increases. It is known that, compared to non-directional sound sources, when the directivity factor is 0, the increase in the reverberation radius is (V/Q-1)ro. Therefore, as the frequency increases, the reverberation radius will further increase ( Because Q>1) at this time. The fact that the seats connected to the orchestra (see Figure 6) always put stringed instruments in front of the orchestra, followed by woodwinds, steel pipes, and triangle bells; acoustically, it is very meaningful. It provides good acoustic conditions for single point pickup.
In addition, the allowable noise level of large-scale natural reverberation music recording studios can be appropriately relaxed. Generally, it is recommended to be less than 25dB(A) or NC-20, and for small ones, it should not be greater than 22dB(A) or NC-15.
These two types of music recording studios are mainly built to meet the requirements of different styles and types of music for the best reverberation time. Therefore, they can both be called "multifunctional music recording studios." This kind of multifunctional recording studio is not realized by sacrificing sound quality requirements and adopting a compromise solution. Therefore, it has certain advantages in terms of economy and production. It is currently one of the more popular forms of music recording studios. With the changes in modern recording technology, especially the sound pickup technology, and the diversification of sound quality processing equipment and technologies, they have attracted more and more attention. Most of the recording studios built or rebuilt after the 1970s in our country take this form, and they can even be used for dialogue recording or mixed recording. This is very economical and practical.
Adjustable reverberation The reverberation time of the music studio is adjusted based on the best reverberation time required by the expected recorded music and its frequency characteristics. It is difficult for this kind of recording studio to meet the expected design requirements, especially when the reverberation adjustment range is large, the acoustic design requirements are higher, but from the perspective of use, it is in parallel with the natural reverberation music recording studio. There is no difference; the natural reverberation plus artificial reverberation type music studio is different. This kind of recording studio takes the optimal reverberation time equal to or less than the expected recording music requirement as the design value of the reverberation time of the recording studio according to its usage requirements. The lack of reverberation is supplemented by artificial reverberation. The biggest feature of this kind of recording studio is that it retains most or part of the natural reverberation components, and can give full play to the freedom of recording production.
In practical terms, in order to give full play to the multiple uses of a room and strive to improve the acoustic conditions of the recording studio, many music recording studios with adjustable reverberation time built at present are a combination of the two basic types mentioned above, that is, in terms of room acoustics conditions , The reverberation time is adjustable, and different degrees of artificial reverberation are often added in the recording production. Most of these recording studios are equipped with sound insulation screens, hooks for hanging reflective surfaces or sound-absorbing surfaces, etc.
The key to the adjustable reverberation studio lies in the area of the absorbing and reflecting surfaces that can be changed. For a room with a certain volume, it is the only factor that determines the adjustable range of the reverberation time. Because the area of the indoor boundary surface is related to the volume of the room, in order to increase the range of adjustable reverberation time, the area of the boundary surface that can be adjusted must be increased. At the same time, in order to meet the needs of large bands, the volume of this type of recording studio is often also bigger. However, because part of the reverberation of this type of recording studio can be provided by artificial reverberation, the indoor reverberation time can be appropriately reduced, and the absorption of the indoor boundary surface is correspondingly increased. The effect is equivalent to increasing the volume of the room. The volume of the room can be appropriately reduced. For example, the music recording studio (room) of China's agricultural film studio built in 1980, although the volume is only 2,860 cubic meters (ground area is about 373 square meters), but due to the use of adjustable reverberation time (reverberation time) The variation range of the question is 0.58-1.40), not only in the case of natural reverberation and artificial reverberation, it is possible to record various types of band performances from a large-scale symphony with 120 people to pop music with a few people, but also Under the conditions of proper use of sound insulation screens and other additional reflective surfaces, dialogue or other language programs can also be recorded, and it can even be used for mixed recording. Fujian Film Studio has proposed mixing in the acoustic design requirements of the continued construction of the recording studio. The response time can be adjusted from 1.50 seconds to 0. The scheme of seven states in 35 seconds. In this way, it is possible to record small orchestral or other types of music and dialogue under natural reverberation conditions in a recording studio with a volume of about 1200 cubic meters (an area of about 175 square meters), and Can be used for mixed recording.
What are the main parameters of a good speaker?
1. Rated impedance. Common rated impedances of speakers are 4 ohms, 6 ohms, 8 ohms, 16 ohms, etc. The impedance of the connected speaker is mostly in the range of 4-16 ohms. The impedance of the speaker should be selected according to the requirements of the power amplifier during use. 2. Effective frequency range. The wider the sound pressure frequency range of the speaker, the better the frequency characteristics. 3. Frequency divider. Generally speaking, the performance of three-way speakers should be better than two-way speakers. Because the three-way frequency adds a mid-range speaker unit, the mid-range can be made more mellow. 4. Sensitivity. A speaker with a sensitivity of 90dB is sufficient to meet the needs of home audio. 5. The caliber of the speaker. The diameter of the woofer is generally 20-38cm, and there are also large diameters of 60cm or 72cm; the diameter of the tweeter is generally 2-6cm, and some are larger than 9cm. 6. The net weight of the speaker. The heavier the speaker, the better the quality. Because the heavier the speaker, the larger the magnet or the thicker the sheet material used in the speaker, both of which will make the sound quality better.
Two operation schemes of digital audio signal compression technology
After the audio signal is digitally encoded, one of the biggest problems facing is the problem of massive data storage and transmission. The compression technology of digital audio signals is a very important link in the digital TV broadcasting system. Compression efficiency and compression quality directly affect the transmission efficiency of digital TV broadcasting and the transmission quality of audio and video. This article mainly analyzes the digital audio compression technology. Compared with analog signals, digital signals have obvious advantages, but digital signals also have their own corresponding disadvantages, that is, the demand for storage capacity and the increase in channel capacity during transmission. Audio compression technology refers to the application of appropriate digital signal processing technology to the original digital audio signal stream (PCM encoding) to reduce (compress) its code rate without loss of useful information or negligible loss. Called compression coding. It must have a corresponding inverse transform, called decompression or decoding. Generally speaking, audio compression techniques can be divided into lossless data compression and lossy data compression. lossless data compression Using a lossless compression scheme can restore the original data information bit by bit after decompression. They eliminate the statistical redundancy that exists in the audio signal by predicting the values in the past samples. A small compression ratio can be achieved, preferably about 2:1, depending on the complexity of the original audio signal. Time-domain predictive coding technology makes lossless compression feasible, thanks to time-domain predictive coding technology. they are: 1. Difference algorithm Audio signals contain repetitive sounds, as well as a lot of redundancy and perceptually irrelevant sounds. Duplicate data information is deleted during the encoding process and re-introduced during decoding. The audio signal is first decomposed into several sub-bands containing discrete tones. Then apply DPCM using a predictor suitable for short-term periodic signals. This kind of coding is adaptive, it looks at the input signal energy to modify the quantization step size. This leads to the so-called adaptive DPCM (ADPCM). 2. Entropy encoder "Using the redundancy in the representation of quantized subband coefficients to improve the efficiency of entropy coding. These coefficients are sent in order of increasing frequency, producing a larger value at low frequencies, and a long stroke close to zero after producing smaller high frequencies. The VLC is taken from a different Huffman table that is most consistent with the statistics of the low-frequency value and the high-frequency value. 3. Block floating point system The binary values from the A/D conversion process are grouped into data blocks, either in the time domain, by using adjacent samples at the A/D conversion transmission output end; or in the frequency domain, by using adjacent samples at the FDCT output end Frequency factor. Then the binary value in the data block is increased proportionally so that the maximum value is only lower than the fully converted value. This conversion factor is called an exponent and is common to all values in the block. Therefore, each value can be determined by a mantissa (a sample value) and an indicator positive number. The bit allocation calculation is derived from the HAS model, and the method to achieve data rate compression is to send the index value once for each data block. The coding performance is good, but the noise is related to the signal content. Shielding technology helps reduce this audible noise. lossy data compression "The way to achieve lossy data compression is to combine two or more processing techniques to take advantage of HAS's inability to detect other high-amplitude specific spectral components. In this way, high-performance data compression schemes and much higher compression ratios from 2:1 to 20:1 can be obtained, depending on the complexity of the encoding/decoding process and audio quality requirements. The lossy data compression system uses perceptual coding technology. The basic principle is to discard all signals below the threshold curve to eliminate the perceptual redundancy in the audio signal. Therefore, these lossy data compression systems are also called perceptually lossless. Perceptually lossless compression is feasible due to the combination of several technologies, such as: 1. Time and frequency domain shielding of signal components. 2. Quantify the noise shielding of each audible tone By allocating enough bits to ensure that the quantization noise level is always lower than the shielding curve. At frequencies close to the audible signal, an SNR of 20 or 30DB is acceptable. 3. Joint coding This technology takes advantage of the redundancy in a multi-channel audio system. It has been
What are the specifications of Bluetooth speakers
At present, there are many Bluetooth headset products on the market, and relatively few Bluetooth speakers. The so-called Bluetooth speaker actually refers to the speaker that relies on the Bluetooth transmission protocol as the carrier for data transmission. Since most mobile devices (mobile phones, notebooks, tablet computers) are equipped with Bluetooth chips, no data cable or audio cable connection is required. It is quickly recognized, easy to operate, and easy to connect. From the perspective of sound quality performance, the effective audio data volume of CD sound quality data (44.1KHz sampling rate, 16bit sampling accuracy) is about 1.4Mbit. To transmit CD sound quality music signals, the transmission rate only needs to be maintained at 2Mbit per second. The "2.1+EDR" specification is sufficient. Moreover, because such products often follow the mature acoustic structure of traditional speakers, and realize wireless playback after integrating the Bluetooth module, their sound quality performance is comparable to that of speaker products of the same level. From the point of view of specifications, although the Bluetooth 3.0/4.0 standard has been proposed, the former is mainly reflected in the Bluetooth radio frequency modulation method in line with Wi-Fi, and the latter is reflected in the application of automatic power control, that is, low power consumption. The two versions reflect the progress of Bluetooth technology, but have little connection with audio applications. From the point of view of chip-level applications, it is suitable for version 3.0/4.0. The mainstream Bluetooth speakers all use the A2DP stereo protocol. In 2012, smartphones, tablets and other devices all support the A2DP protocol, so there is no obstacle to the use of Bluetooth speakers.
Does the speaker have to be burned?
In the first stage, it is only a warm-up stage, and it is not necessary to select the tracks from the recommended reference tracks above. Just use a more soothing tune like "Guess the Heart" for normal playback at a volume of about 30% of the normal volume, and the playback time is generally 10 to 12 hours. After the first stage of warm-up, even if it enters the official appraisal stage, there are clear standards for the selection of music types and styles, down to the adjustment of volume and playback time period. First of all, from the three types of high, middle and low audio of the recommended reference tracks, Liang Zhu, Dukou, and Xianyun Guhe were selected as the praise machine tracks. Use a volume of 60% to 70% of the normal volume for a 48-hour loop playback. This stage is a link between the previous and the next. The adaptive memory of the headset for each frequency is formed at this stage. The success or failure of this stage will directly affect the performance of the headset for the sound of the frequency used in the future work process. Of course, in addition to the three recommended songs above, you can also use other songs, such as Four Seasons Autumn, Qinghai-Tibet Plateau, etc. mentioned in the table. The third stage is mainly a consolidation stage. The main role played is to further enhance the return visit effect of the corresponding frequency band, so that the corresponding frequency band can be better interpreted in the future playback. Here, the trial comprehensive performance is superb. Strong California Hotel, The mass (Era), Fairytale (Secret Garden) three songs. Need to pay attention to when playing. Keep the volume at about 50% for loop playback, and the playback time required is about 14 hours. The whole testimonial process is over here. The rest is the adaptive stage of ordinary music. Under normal circumstances, after about two weeks, the performance of the headphones will feel completely reborn. Remember: Do not use rock and dance music, so the speakers will be useless!