| STAGE TECHNIC/Pro Sound Author:tuyenphuc Phần 5: Chất lượng âm thanh (Sound quanlity). How to Prevent Distortion Unwanted distortion is caused by a signal which is "too strong". If an audio signal level is too high for a particular component to cope with, then parts of the signal will be lost. This results in the rasping distorted sound. To illustrate this point, the pictures below represent a few seconds of music which has been recorded by a digital audio program. The maximum possible dynamic range (the range from quietest to loudest parts) of the signal is shown as 0 to +/-100 units. In the first example, the amplitude (strength / height) of the signal falls comfortably within the +/-100 unit range. This is a well-recorded signal. In the second example, the signal is amplified by 250%. In this case, the recording components can no longer accommodate the dynamic range, and the strongest portions of the signal are cut off. This is where distortion occurs. These examples can be used as an analogy for any audio signal. Imagine that the windows above represent a pathway through a component in a sound system, and the waves represent the signal travelling along the pathway. Once the component's maximum dynamic range is breached, you have distortion. Minimising Distortion Distortion can occur at almost any point in the audio pathway, from the microphone to the speaker. The first priority is to find out exactly where the problem is. Ideally, you would want to measure the signal level at as many points as possible, using a VU (Volume Unit) meter or similar device. Generally speaking, you should keep the level below about 0dBu at every point in the pathway. If you can't measure the signal level, you'll have to do some deducing. Follow the entire audio pathway, beginning at the source (the source could be a microphone, tape deck, musical instrument, etc). Here are some things to look for: Is the distortion coming from a microphone? This could be caused by a very loud noise being too close to the mic. Try moving the mic further away from the noise source. Are you seeing any "peak" or "clip" lights on any of your equipment? These are warnings that a signal level is too high. Are any volume or gain controls in your system turned up suspiciously high? Are there any obvious points where you could drop the level? Are your speakers being driven too hard? If you have an amplifier which is pushing the speakers beyond their design limits, then be careful -� you may well find that the distortion becomes permanent. If the distortion is coming from occasional peaking, consider adding a compressor. Could the distortion be caused by faulty equipment? Is the problem really distorion? There are some other unpleasant noises which could be confused with distorion; for example, the graunching sounds made by a dodgy cable connection or dirty volume knob. How to Eliminate Feedback Audio feedback is the ringing noise (often described as squealing, screeching, etc) sometimes present in sound systems. It is caused by a "looped signal", that is, a signal which travels in a continuous loop. In technical terms, feedback occurs when the gain in the signal loop reaches "unity" (0dB gain). One of the most common feedback situations is shown in the diagram below - a microphone feeds a signal into a sound system, which then amplifies and outputs the signal from a speaker, which is picked up again by the microphone. Of course, there are many situations which result in feedback. For example, the microphone could be replaced by the pickups of an electric guitar. (In fact many guitarists employ controlled feedback to artistic advantage. This is what's happening when you see a guitarist hold his/her guitar up close to a speaker.) To eliminate feedback, you must interrupt the feedback loop. Here are a few suggestions for controlling feedback: Change the position of the microphone and/or speaker so that the speaker output isn't feeding directly into the mic. Keep speakers further forward (i.e. closer to the audience) than microphones. Use a more directional microphone. Speak (or sing) close to the microphone. Turn the microphone off when not in use. Equalise the signal, lowering the frequencies which are causing the feedback. Use a noise gate (automatically shuts off a signal when it gets below a certain threshold) or filter. Lower the speaker output, so the mic doesn't pick it up. Avoid aiming speakers directly at reflective surfaces such as walls. Use direct injection feeds instead of microphones for musical instruments. Use headset or in-ear monitors instead of speaker monitors. You could also try a digital feedback eliminator. There are various models available with varying levels of effectiveness. The better ones are reported to produce reasonable results. Other Notes: Feedback can occur at any frequency. The frequencies which cause most trouble will depend on the situation but factors include the room's resonant frequencies, frequency response of microphones, characteristics of musical instruments (e.g. resonant frequencies of an acoustic guitar), etc. Feedback can be "almost there", or intermittent. For example, you might turn down the level of a microphone to stop the continuous feedback, but when someone talks into it you might still notice a faint ringing or unpleasant tone to the voice. In this case, the feedback is still a problem and further action must be taken. Audio Equalization Equalization, or EQ for short, means boosting or reducing (attenuating) the levels of different frequencies in a signal. The most basic type of equalization familiar to most people is the treble/bass control on home audio equipment. The treble control adjusts high frequencies, the bass control adjusts low frequencies. This is adequate for very rudimentary adjustments — it only provides two controls for the entire frequency spectrum, so each control adjusts a fairly wide range of frequencies. Advanced equalization systems provide a fine level of frequency control. The key is to be able to adjust a narrower range of frequencies without affecting neighbouring frequencies. Equalization is most commonly used to correct signals which sound unnatural. For example, if a sound was recorded in a room which accentuates high frequencies, an equalizer can reduce those frequencies to a more normal level. Equalization can also be used for applications such as making sounds more intelligible and reducing feedback. There are several common types of equalization, described below. Shelving EQ In shelving equalization, all frequencies above or below a certain point are boosted or attenuated the same amount. This creates a "shelf" in the frequency spectrum. Bell EQ Bell equalization boosts or attenuates a range of frequencies centred around a certain point. The specified point is affected the most, frequencies further from the point are affected less. Graphic EQ Graphic equalizers provide a very intuitive way to work — separate slider controls for different frequencies are laid out in a way which represents the frequency spectrum. Each slider adjusts one frequency band so the more sliders you have, the more control. A graphic equalizer is, as the name implies, an equalizer which uses a graphical layout to represent the changes made. It uses a series of sliders (usually vertical) which correspond to a set of fixed frequency bands. You raise or lower each slider to boost or lower (attenuate) the level of that frequency band. Graphic equalizers are commonly referred to by the number of bands (e.g. 51-band, 31-band) or by the frequency separation of each band expressed in octaves (e.g. 2/3 octave, 1/3 octave, 1/6 octave). Parametric EQ Parametric equalizers use bell equalization, usually with knobs for different frequencies, but have the significant advantage of being able to select which frequency is being adjusted. Parametrics are found on sound mixing consoles and some amplifier units (guitar amps, small PA amps, etc). Parametric Equalizers The word parametric means something which has one or more parameters on which the outcome depends. When applied to audio equalization, this means equalization which depends on parameters such as centre frequency, bandwidth and amplitude. The user is able to adjust these parameters to determine exactly how the equalization is applied. Centre Frequency The most important feature of a parametric equaliser is that it allows you to select which frequency to adjust. For example, instead of having a simple mid-range adjustment which boosts or reduces a pre-set range of frequencies, you can specify exactly which mid-range frequency to boost or reduce. This gives you great flexibility and accuracy. The illustration on the right shows parametric controls for upper-mid-range frequencies. These controls work together — the brown knob determines which frequency is to be adjusted (0.6KHz to 10KHz) and the green knob makes the adjustment (-15dB to +15dB). Note that although you select a specific frequency, the actual adjustment will apply to frequencies above and below this frequency as well. This is why it is called the centre frequency — it is the frequency at the centre of the adjustment. Example of use: Let's say you have a feedback problem somewhere in the 5KHz range but you aren't sure of the exact frequency. Turn the green knob right down, then slowly rotate the brown knob through the frequency range. As you do so, you will hear the selected frequencies being reduced. When you reach the frequency which is causing the feedback, the feedback will be reduced. Bandwidth Control (Q) As noted above, adjustments are made to a range of frequencies around the centre frequency. The bandwidth control determines how far above and below the centre frequency the adjustment will affect, i.e. the width or spread of frequencies. A narrow bandwidth adjustment is very specific, useful for accurately removing or accentuating a specific frequency. This would be helpful in the feedback situation described above — once you have identified the offending frequency, reduce the bandwidth so you are adjusting the smallest range possible while still eliminating the feedback. A broader bandwidth affects more frequencies, useful for adjusting a wider range such as the upper frequencies in a voice. Broader adjustments tend to sound more natural. Note: Bandwidth controls are not available on all parametric equalizers. Amplitude The is the level of adjustment, measured in decibels (dB). Phần 6: -Audio monitoring & metering. -Thiết bị xử lý tín hiệu (processing). -Kỹ xảo (effect). Audio Monitoring & Metering Audio Metering means using a visual display to monitor audio levels. This helps maintain audio signals at their optimum level and minimise degradation. There are two common types of meter which are used to measure audio levels: VU Meter (Volume Unit) PPM Meter (Peak Program) Both types of meter are available in various forms including stand-alone units, components in larger systems, and software applications. Whatever the type of meter, two characteristics are important: The scale which defines which units are being measured. The ballistics of the meter which determine how fast it responds to sound and returns to a lower level. VU Meter A VU (volume unit) meter is an audio metering device. It is designed to visually measure the "loudness" of an audio signal. The VU meter was developed in the late 1930s to help standardise transmissions over telephone lines. It went on to become a standard metering tool throughout the audio industry. VU meters measure average sound levels and are designed to represent the way human ears perceive volume. The rise time of a VU meter (the time it takes to register the level of a sound) and the fall time (the time it takes to return to a lower reading) are both 300 milliseconds. The optimum audio level for a VU meter is generally around 0VU, often referred to as "0dB". Technically speaking, 0VU is equal to +4 dBm, or 1.228 volts RMS across a 600 ohm load. VU meters work well with continuous sounds but poorly with fast transient sounds. Peak Program Meter (PPM) A Peak Program Monitor (PPM), sometimes referred to as a Peak Reading Meter (PRM), is an audio metering device. It's general function is similar to a VU meter but there are some important differences. The rise time of a PPM (the time it takes to register the level of a sound) is much faster than a VU meter, typically 10 milliseconds compared to 300 milliseconds. This makes transient peaks easier to measure. The fall time of a PPM (the time it takes the meter to return to a lower reading) is much slower. PPM meters are very good for reading fast, transient sounds. This is especially useful in situations where pops and distortion are a problem. Audio Compression Audio compression is a method of reducing the dynamic range of a signal. All signal levels above the specified threshold are reduced by the specified ratio. The example below shows how a signal level is reduced by 2:1 (the output level above the threshold is halved) and 10:1 (severe compression). How to Use a Compressor Audio compression is a method of reducing the dynamic range of a signal. You will need: A compressor with manual controls. An audio source to be compressed (eg. microphone, musical instrument, output of sound desk, etc). A destination device with which to feed the compressed output (eg. tape deck, sound desk, amplifier, etc). Connect the source to the compressor's input, and the compressor's output to the destination device. Adjust the compressor's input and output gains to appropriate levels. Set the threshold level to the point at which you wish compression to take effect. Signals below this level will not be affected. Signal levels above the threshold will be reduced according to the compression ratio. Set the compression ratio. Ratios of 5:1 or less will produce fairly smooth compression; ratios of 10:1 or more will produce more severe cutting off. Set the attack time. This is the delay between detection of a signal above the threshold, and the commencement of compression (ie. the time it takes to "attack" the signal). Set the decay time. This is the time taken to release the signal from compression. Adjust any other settings on the compressor. If you don't know what they are, try to put them on automatic, or disable them. Example: Set the compressor to a threshold of 0db, and a compression ratio of 3:1. In this case, all signals below 0db will be unaffected, and all signals above 0db will be reduced by 3db to 1 (ie. for every 1db input over 0db, 1/3db will be output). Audio Limiters A limiter is a type of compressor designed for a specific purpose — to limit the level of a signal to a certain threshold. Whereas a compressor will begin smoothly reducing the gain above the threshold, a limiter will almost completely prevent any additional gain above the threshold. A limiter is like a compressor set to a very high compression ratio (at least 10:1, more commonly 20:1 or more). The graph below shows a limiting ratio of infinity to one, i.e. there is no gain at all above a the threshold. Input Level vs Output Level With Limiting Threshold Limiters are used as a safeguard against signal peaking (clipping). They prevent occasional signal peaks which would be too loud or distorted. Limiters are often used in conjunction with a compressor — the compressor provides a smooth roll-off of higher levels and the limiter provides a final safety net against very strong peaks. Audio Expansion Audio expansion means to expand the dynamic range of a signal. It is basically the opposite of audio compression. Like compressors and limiters, an audio expander has an adjustable threshold and ratio. Whereas compression and limiting take effect whenever the signal goes above the threshold, expansion effects signal levels below the threshold. Any signal below the threshold is expanded downwards by the specified ratio. For example, if the ratio is 2:1 and the signal drops 3dB below the threshold, the signal level will be reduced to 6dB below the threshold. The following graph illustrates two different expansion ratios — 2:1 and the more severe 10:1. Input Level vs Output Level With Expansion An extreme form of expander is the noise gate, in which lower signal levels are reduced severely or eliminated altogether. A ratio of 10:1 or higher can be considered a noise gate. Note: Some people also use the term audio expansion to refer to the process of decompressing previously-compressed audio data. Audio Effects This page provides an overview of the most common audio effects used in sound production, with links to more detailed tutorials. Equalization Equalization means boosting or reducing (attenuating) the levels of various frequencies in a signal. At it's most basic, equalization can mean turning the bass/treble controls up or down. Advanced equalizers have fine controls for specific frequencies. Common uses for equalization include correct signals which sound unnatural and reducing feedback. Compression & Limiting Compression means reducing the dynamic range of a signal. All signal values above a certain adjustable threshold are reduced in gain relative to lower-level signals. This creates a more even signal level, reducing the level of the loudest parts. Limiting is an extreme form of compression. Rather than smoothly reducing the gain of successively higher levels, all signal above the threshold is limited to the same gain. This creates a very hard cut-off point, over which there is no increase in level. Expansion & Noise Gating Expansion means increasing the dynamic range of a signal. High level signals maintain the same (or nearly the same) levels, low level signals are reduced (attenuated). This creates a greater range between quiet and loud. Expansion is the opposite of compression. Noise gating is an extreme form of expansion — signals below a certain point are either heavily attenuated or eliminated completely. This leaves only higher level signals and removes background noise when the signal is not present. Delay / Echo Delay is a simple concept — the original audio signal is followed closely by a delayed repeat, just like an echo. The delay time can be as short as a few milliseconds or as long as several seconds. A delay effect can include a single echo or multiple echoes, usually reducing quickly in relative level. Delay also forms the basis of other effects such as reverb, chorus, phasing and flanging. Reverb (Reverberation) Reverb is short for reverberation, the effect of many sound reflections occurring in a very short space of time. The familiar sound of clapping in an empty hall is a good example of reverb. Reverb effects are used to restore the natural ambience to a sound, or to give it more fullness and body. What is Reverb? Reverberation, or reverb for short, refers to the way sound waves reflect off various surfaces before reaching the listener's ear. The example on the right shows one person (the sound source) speaking to another person in a small room. Although the sound is projected most strongly toward the listener, sound waves also project in other directions and bounce off the walls before reaching the listener. Sound waves can bounce backwards and forwards many times before they die out. When sound waves reflect off walls, two things happen: They take longer to reach the listener. They lose energy (get quieter) with every bounce. The listener hears the initial sound directly from the source followed by the reflected waves. The reflections are essentially a series of very fast echoes, although to be accurate, the term "echo" usually means a distinct and separate delayed sound. The echoes in reverberation are merged together so that the listener interprets reverb as a single effect. In most rooms the reflected waves will scatter and be absorbed very quickly. People are seldom consciously aware of reverb, but subconsciously we all know the difference between "inside sound" and "outside sound". Outside locations, of course, have no walls and virtually no reverb unless you happen to be close to reflective surfaces. Some rooms result in more reverb than others. The obvious example is a hall with large, smooth reflective walls. When the hall is empty, reverb is most pronounced. When the hall is full of people, they absorb a lot of sound waves so reverb is reduced. Reverb Effects Reverberation can be added to a sound artificially using a reverb effect. This effect can be generated by a stand-alone reverb unit, the reverb effect in another device (such as a mixer or multi-effects unit), or by audio processing software. There are three possible reasons for adding reverb: To restore the natural sound as the listener would expect to hear it. For example, a recording done in a very low-reverb studio might sound unnatural unless reverb is added. To enhance the sound. For example, it is common to give vocal recordings more reverb than what would be considered natural. Reverb helps fill out the voice, giving it more "body" and is usually considered to be a flattering effect. Reverb can even help smooth minor vocal fluctuations so they aren't as obvious. To create special effects such as dream sequences, etc. Reverb is the most common audio effect, partly because it is used in so many situations from music studios to television production. Every sound operator should have a good understanding of reverb and how/when to apply it. It pays to be judicious with reverb. Because it is so effective, it can easily be over-used. The right amount of reverb can do wonders for a singer's voice but too much sounds silly. Examples The photo below is a rack-mountable Lexicon PCM 81 Digital Effects Processor. This unit has a number of effects including reverb. The screenshot below is from Adobe Audition, a sound editing package. It gives you an idea of some of the common reverb settings. Notice how most of the presets are described by the real-world effect they are simulating, for example, "Concert Hall" and "Medium Empty Room". This is common in reverb units. Examples of Reverb The following examples show how the reverb effect works. The first example is dry, meaning that it has no effects or other processing applied. The next two examples have different levels of reverb applied. Drums - Dry Drums - Medium Reverb Drums - Hall Chorus The chorus effect is designed to make a signal sound like it was produced by multiple similar sources. For example, if you add the chorus effect to a solo singer's voice, the results sounds like.... a chorus. Chorus works by adding multiple short delays to the signal, but rather than repeating the same delay, each delay is "variable length" (the speed and length of the delay changes). This adds the randomness required for the chorus sound. Varying the delay time also varies the pitch slightly, further adding to the "multiple sources" illusion. The chorus effect was originally designed to make a single person's voice sound like multiple voices saying or singing the same thing, i.e. make a soloist into a chorus. It has since become a common effect used with musical instruments as well. The effect is a type of delay — the original signal is duplicated and played at varying lengths and pitches. This creates the effect of multiple sources, as each source is very slightly out of time and tune (just as in real life). Technically, a chorus is similar to a flanger. Common parameters include: Number of Voices: The number of times the source is multiplied. Delay: The minimum delay length, typically 20 to 30 milliseconds. Sweep Depth/ Width: The maximum delay length. The following example is the chorus settings window in Adobe Audition. Phasing & Flanging Phasing, AKA phase shifting, is a sweeping, whooshing effect often used in music. The effect is created by mixing the original signal with another version of itself which has been phase-shifted. This results in various out-of-phase interactions over time which gives the sweeping effect. Phasing is created by adding evenly-spaced notches in the frequency response and moving them up and down the frequency spectrum. Flanging is a specific type of phasing which uses notches that are "harmonically related", i.e. related to musical notes. Phase Shifting (Phasing) Phase-shifting, AKA phasing, is an audio effect which takes advantage of the way sound waves interact with each other when they are out of phase. By splitting an audio signal into two signals and changing the relative phasing between them, a variety of interesting sweeping effects can be created. The phasing effect was first made popular by musicians in the 1960s and has remained an important part of audio work ever since. Phasing is similar to flanging, except that instead of a simple delay it uses notch and boost filters to to phase-shift frequencies over time. The following examples show some of the different types of phasing effects (MP3): Drums: Dry (original audio with no effect) Drums: Phased Drums: Crunchy Phase Drums: Trebly Phasing Drums: Bassy Phase Drums: Tremolo Phasing Left to Right Drums: Washy Phase Left to Right Drums: "Bubbles" Phase The screenshot below is from Adobe Audition and shows some of the common settings available in phasing effects. Flanging Flanging is a type of phase-shifting. It is an effect which mixes the original signal with a varying, slightly delayed version of the signal. The original and delayed signals are mixed more or less equally. Flanging results in a sweeping sound — see the following example (MP3): Drums: Dry (original audio with no effect) Drums: Flanged The term flanging comes from the days of reel-to-reel tape recording. The original signal was recorded on a second reel, and the delay was achieved by holding a finger or thumb on the edge (flange) of the reel to physically slow it down. Flanging was made popular during the psychedelic music era in the 1960s and 1970s. The following example is the flanger settings window in Adobe Audition. It shows some of the settings commonly used in flanging: (Xem tiếp phần 7: Màu sắc âm thanh, Noise, Colours & Types). |
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STAGE TECHNIC/Pro Sound Author:tuyenphuc Phần 3: Cân bằng trong âm thanh. Balanced Audio This tutorial explains how balanced audio systems work. It is suitable for people who have a basic understanding of audio cables and connectors, as well as simple wave interactions (such as how waves from different sources interfere with each other). If you don't understand these things, take our introduction to audio tutorial first. What is Balanced Audio? Balanced audio is a method of minimizing unwanted noise from interference in audio cables. The idea is that any interference picked up in a balanced cable is eliminated at the point where the cable plugs into a sound mixer or other equipment. Balanced audio works on the principle that two identical signals which are inverted 180° out of phase will cancel each other out. The cables used in such systems are designed to carry two versions of the signal and manipulate the relative phases of these signals to eliminate noise. This will make more sense when we look at how balanced cables work, but first we need to take a step backwards and look at unbalanced audio cables. Unbalanced Audio Cables Traditional unbalanced cables use two lines to transmit the audio signal - a hot line which carries the signal and an earth line. This is all that is required to transmit audio and is common in short cables (where noise is less of a problem) and less professional applications. � Note: Internal componentry (in sound mixers etc) is also unbalanced. Unbalanced Audio Connectors Unbalanced audio cables are commonly associated with the 1/4" phono jack connector and the RCA connector. However any single-pin connector used for audio is unbalanced. 3-pin XLRs can also be used for unbalanced cables. For more information about these connectors, including how to wire them, see Audio Connections. 1/4" phono RCA ������ � � � Balanced Audio Cables Balanced audio cables use an extra line, and consist of a hot line (positive), cold line (negative) and earth. The audio signal is transmitted on both the hot and cold lines, but the voltage in the cold line is inverted so it is negative when the hot signal is positive. These two signals are often referred to as being 180 degrees out of phase with each other. This terminology can be confusing — it does not mean one signal is delayed until it is out of phase, it means one signal is effectively flipped upside down. � When the cable is plugged into an input (on a mixer or other equipment) the hot and cold signals are combined. Normally you would expect these two signals to cancel each other out, but at the input stage they are put "back in phase" (i.e. the inversion is reversed) before being merged together, so they actually combine to form a stronger signal. Removing Noise Along the length of the cable, noise can be introduced from external sources such as power cables, RF interference, etc. This noise will be identical on both hot and cold lines. This is known as a common mode signal - a signal which appears equally on both conductors of a two wire line. So the hot and cold lines carry two signals: A desirable audio signal which has an opposite voltage on each line, and unwanted noise which is the same on both lines. This is where the trick of balanced audio kicks in. At the input stage when the inverted audio signal is re-inverted to make both desirable audio signals the same, the unwanted noise is inverted (i.e. put out of phase). Viola - all the unwanted noise is cancelled out, leaving only the combined original signal. Combining Balanced Cables The standard connector for balanced audio is the 3-pin XLR. For details on wiring various configurations and connectors see Audio Connections. Unfortunately there is no official standard for wiring balanced audio cables, but the most common configuration is: Pin 1: Shield (Ground) Pin 2: Hot Pin 3: Cold Mixing Wiring Configurations Using cables or equipment with different wiring configurations in the same system is a recipe for trouble. You may well find that audio signals start canceling each other out and leave you with nothing. Many sound mixers have a "phase invert" switch on each channel. This swaps the phasing of the hot and cold pins to solve the mismatch problem. Obviously the best plan is to keep your wiring consistent. Use the configuration above and you shouldn't experience too many problems. Last Word The rule of thumb for audio systems is: Connect all shields, ground everything, and balance wherever possible. Phần 4: Sound Mixer. Sound Mixers: Overview A sound mixer is a device which takes two or more audio signals, mixes them together and provides one or more output signals. The diagram on the right shows a simple mixer with six inputs and two outputs. As well as combining signals, mixers allow you to adjust levels, enhance sound with equalization and effects, create monitor feeds, record various mixes, etc. Mixers come in a wide variety of sizes and designs, from small portable units to massive studio consoles. The term mixer can refer to any type of sound mixer; the terms sound desk and sound console refer to mixers which sit on a desk surface as in a studio setting. Sound mixers can look very intimidating to the newbie because they have so many buttons and other controls. However, once you understand how they work you realize that many of these controls are duplicated and it's not as difficult as it first seems. Applications Some of the most common uses for sound mixers include: Music studios and live performances: Combining different instruments into a stereo master mix and additional monitoring mixes. Television studios: Combining sound from microphones, tape machines and other sources. Field shoots: Combining multiple microphones into 2 or 4 channels for easier recording. Channels Mixers are frequently described by the number of channels they have. For example, a "12-channel mixer" has 12 input channels, i.e. you can plug in 12 separate input sources. You might also see a specification such as "24x4x2" which means 24 input channels, 4 subgroup channels and two output channels. More channels means more flexibility, so more channels is generally better. See mixer channels for more information. Advanced Mixing The diagram below shows how a mixer can provide additional outputs for monitoring, recording, etc. Even this is just scratching the surface of what advanced mixers are capable of. � Sound Mixer: Channels Each input source comes into the mixer through a channel. The more channels a mixer has, the more sources it can accept. The following examples show some common ways to describe a mixer's compliment of channels: 12-channel 12 input channels. 16x2 16 input channels, 2 output channels. 24x4x2 24 input channels, 4 subgroup channels and two output channels. Input Channels On most sound desks, input channels take up most of the space. All those rows of knobs are channels. Exactly what controls each channel has depends on the mixer but most mixers share common features. The list below details the controls available on a typical mixer channel. Input Gain / Attenuation: The level of the signal as it enters the channel. In most cases this will be a pot (potentiometer) knob which adjusts the level. The idea is to adjust the levels of all input sources (which will be different depending on the type of source) to an ideal level for the mixer. There may also be a switch or pad which will increase or decrease the level by a set amount (e.g. mic/line switch). Phantom Power: Turns phantom power on or off for the channel. Equalization: Most mixers have at least two EQ controls (high and low frequencies). Good mixers have more advanced controls, in particular, parametric equalization. Auxiliary Channels: Sometimes called aux channels for short, auxiliary channels are a way to send a "copy" of the channel signal somewhere else. There are many reasons to do this, most commonly to provide separate monitor feeds or to add effects (reverb etc). Pan & Assignment: Each channel can be panned left or right on the master mix. Advanced mixers also allow the channel to be "assigned" in various ways, e.g. sent directly to the main mix or sent only to a particular subgroup. Solo / Mute / PFL: These switches control how the channel is monitored. They do not affect the actual output of the channel. Channel On / Off: Turns the entire channel on or off. Slider: The level of the channel signal as it leaves the channel and heads to the next stage (subgroup or master mix). Subgroup Channels Larger sound desks usually have a set of subgroups, which provide a way to sub-mix groups of channels before they are sent to the main output mix. For example, you might have 10 input channels for the drum mics which are assigned to 2 subgroup channels, which in turn are assigned to the master mix. This way you only need to adjust the two subgroup sliders to adjust the level of the entire drum kit. Sound Mixers: Channel Inputs The first point of each channel's pathway is the input socket, where the sound source plugs into the mixer. It is important to note what type of input sockets are available — the most common types are XLR, 6.5mm Jack and RCA. Input sockets are usually located either on the rear panel of the mixer or on the top above each channel. � There are no hard-and-fast rules about what type of equipment uses each type of connector, but here are some general guidelines: XLR Microphones and some audio devices. Usually balanced audio, but XLR can also accommodate unbalanced signals. 6.5mm Jack Musical instruments such as electric guitars, as well as various audio devices. Mono jacks are unbalanced, stereo jacks can be either unbalanced stereo or balanced mono. RCA Musical devices such as disc players, effects units, etc. Input Levels The level of an audio signal refers to the voltage level of the signal. Signals can be divided into three categories: Mic-level (low), line-level (a bit higher) and loudspeaker-level (very high). Microphones produce a mic-level signal, whereas most audio devices such as disc players produce a line-level signal. Loudspeaker-level signals are produced by amplifiers and are only appropriate for plugging into a speaker — never plug a loudspeaker-level signal into anything else. Sound mixers must be able to accommodate both mic-level and line-level signals. In some cases there are two separate inputs for each channel and you select the appropriate one. It is also common to include some sort of switch to select between inputs and/or signal levels. Input Sockets and Controls The example on the right shows the input connections on a typical mixer. This mixer has two input sockets — an XLR for mic-level inputs and a 6.5mm jack for line-level inputs. It also has a pad button which reduces the input level (gain) by 20dB. This is useful when you have a line-level source that you want to plug into the mic input. Some mixers also offer RCA inputs or digital audio inputs for each channel. Some mixers provide different sockets for different channels, for example, XLR for the first 6 channels and RCA for the remainder. Input Gain When a signal enters the mixer, one of the first controls is the input gain. This is a knob which adjusts the signal level before it continues to the main parts of the channel. The input gain is usually set once when the source is plugged in and left at the same level — any volume adjustments are made by the channel fader rather than the gain control. Set the gain control so that when the fader is at 0dB the signal is peaking around 0dB on the VU meters. Other Controls and Considerations Phasing: Some equipment and cables are wired with different phasing, that is, the wires in the cable which carry the signal are arranged differently. This will kill any sound from that source. To fix this problem, some mixers have a phase selector which will change the phasing at the input stage. Phantom Power: Some mixers have the option to provide a small voltage back up the input cable to power a microphone or other device. See Phantom Power for more information. Phantom Power Phantom power is a means of distributing a DC current through audio cables to provide power for microphones and other equipment. The supplied voltage is usually between 12 and 48 Volts, with 48V being the most common. Individual microphones draw as much current from this voltage as they need. A balanced audio signal connected to a 3 pin XLR has the audio signal traveling on the two wires – usually connected to pin 2 (+ve) and pin 3 (-ve). Pin 1 is connected to the shield, which is earthed. The audio signal is an AC (alternating current), whereas phantom power is DC (direct current). The DC phantom power is transmitted simultaneously on both pin 2 and 3, with the shield (pin 1) being the return path. Since the DC voltage on the ‘hot’ and ‘cold’ pins (2 & 3) is identical, it is seen by equipment as “common mode” noise and rejected, or ignored, by the equipment. If you put a volt meter on pins 1 & 2, or pins 1 & 3, you will see the 48v DC phantom power, but if you meter pins 2 & 3 (the audio carrying wires) you will see no voltage. The DC voltage can be harnessed however, and used to power mics, mic-line amps, or indeed a video camera (in this case the DC voltage would travel up the video cable – and would need special equipment to filter this voltage). Phantom powering is defined in DIN standard 45 596 or IEC standard 268-15A Note: Audio signals transmit as AC current, whereas powered equipment requires DC current to operate. Phantom power is a clever way of using one cable to transmit both currents. How is Phantom Power Generated? Phantom power can be generated from sound equipment such as mixing consoles and preamplifiers. Special phantom power supplies are also available. Does Phantom Power Affect the Audio? No, it does not affect the quality of audio at all and is quite safe to use. However it is recommended that you do not supply phantom power to microphones which do not require it, especially ribbon microphones. Sound Mixers: Channel Equalization Most mixers have some of sort equalization controls for each channel. Channel equalizers use knobs (rather than sliders), and can be anything from simple tone controls to multiple parametric controls. The first example on the right is a simple 2-way equalizer, sometimes referred to as bass/treble or low/high. The upper knob adjusts high frequencies (treble) and the lower knob adjusts low frequencies (bass). This is a fairly coarse type of equalization, suitable for making rough adjustments to the overall tone but is not much use for fine control. � This next example is a 4-way equalizer. The top and bottom knobs are simple high and low frequency adjustments (HF and LF). The middle controls consist of two pairs of knobs. These pairs are parametric equalizers — each pair works together to adjust a frequency range chosen by the operator. The brown knob selects the frequency range to adjust and the green knob makes the adjustment. The top pair works in the high-mid frequency range (0.6KHz to 10KHz), the lower pair works in the low-mid range (0.15 to 2.4KHz). The "EQ" button below the controls turns the equalization on and off for this channel. This lets you easily compare the treated and untreated sound. It is common for mixers with parametric equalizers to combine each pair of knobs into a single 2-stage knob with one on top of the other. This saves space which is always a bonus for mixing consoles. Notes about Channel Equalization If the mixer provides good parametric equalization you will usually find that these controls� �are more than adequate for equalizing individual sources. If the mixer is limited to very simple equalization, you may want to use external equalizers. For example, you could add a graphic equalizer to a channel using the insert feature. In many situations you will use additional equalization outside the mixer. In live sound situations, for example, you will probably have at least one stereo graphic equalizer on the master output. Sound Mixers: Auxiliary Channels Most sound desks include one or more auxiliary channels (often referred to as aux channelsinput channel's audio signal to another destination, independent of the channel's main output. for short). This feature allows you to send a secondary feed of an The example below shows a four-channel mixer, with the main signal paths shown in green. Each input channel includes an auxiliary channel control knob — this adjusts the level of the signal sent to the auxiliary output (shown in blue). The auxiliary output is the sum of the signals sent from each channel. If a particular channel's auxiliary knob is turned right down, that channel is not contributing to the auxiliary channel. � � In the example above, the auxiliary output is sent to a monitoring system. This enables a monitor feed which is different to the main output, which can be very useful. There are many other applications for auxiliary channels, including: Multiple separate monitor feeds. Private communication, e.g. between the sound desk and the stage. Incorporating effects. Recording different mixes. Mixers are not limited to a single auxiliary channel, in fact it is common to have up to four or more. The following example has two auxiliary channels — "Aux 1" is used for a monitor and "Aux 2" is used for an effects unit. � Note that the monitor channel (Aux 1) is "one way", i.e. the channel is sent away from the mixer and doesn't come back. However the Aux 2 channel leaves the mixer via the aux sendaux return input. It is then mixed into the master stereo bus. output, goes through the effects unit, then comes back into the mixer via the Pre / Post Fader The auxiliary output from each channel can be either pre-fader or post-fader. A pre-fader output is independent of the channel fader, i.e. the auxiliary output stays the same level whatever the fader is set to. A post-fader output is dependent on the fader level. If you turn the fader down the auxiliary output goes down as well. Many mixers allow you to choose which method to use with a selector button. The example pictured right shows a mixer channel with four auxiliary channels and two pre/post selectors. Each selector applies to the two channels above it, so for example, the button in the middle makes both Aux 1 and Aux 2 either pre-fader or post-fader. Sound Mixers: Channel Assigning & Panning One of the last sets of controls on each channel, usually just before the fader, is the channel assign and pan. Pan Almost all stereo mixers allow you to assign the amount of panning. This is a knob which goes from full left to full right. This is where the channel signal appears on the master mix (or across two subgroups if this is how the channel is assigned). If the knob is turned fully left, the channel audio will only come through the left speaker in the final mix. Turn the knob right to place the channel on the right side of the mix. Assign This option may be absent on smaller mixers but is quite important on large consoles. The assign buttons determine where the channel signal is sent. In many situations the signal is simply sent to the main master output. In small mixers with no assign controls this happens automatically. However you may not want a channel to be fed directly into the main mix. The most common alternative is to send the channel to a subgroup first. For example, you could send all the drum microphones to their own dedicated subgroup which is then sent to the main mix. This way, you can adjust the overall level of all the drums by adjusting the subgroup level. In the example pictured right, the options are: Mix: The channel goes straight to the main stereo mix 1-2: The channel goes to subgroup 1 and/or 2. If the pan control is set fully left the channel goes only to subgroup 1, if the pan is set fully right the channel goes only to subgroup 2. If the pan is centered the channel goes to subgroups 1 and 2 equally. 3-4: The channel goes to subgroups 3 and/or 4, with the same conditions as above. For stereo applications it is common to use subgroups in pairs to maintain stereo separation. For example, it is preferable to use two subgroups for the drums so you can pan the toms and cymbals from left to right. You can assign the channel to any combination of the available options. In some cases you may not want the channel to go to the main mix at all. For example, you may have a channel set up for communicating with the stage via an aux channel. In this case you don't assign the channel anywhere. Sound Mixers: PFL PFL means Pre-Fade Listen. It's function is to do exactly that — listen to the channel's audio at a point before the fader takes effect. The PFL button is usually located just above the channel fader. In the example on the right, it's the red button (the red LED lights when PFL is engaged). Note: PFL is often pronounced "piffel". When you press the PFL button, the main monitor output will stop monitoring anything else and the only audio will be the selected PFL channel(s). This does not affect the main output mix — just the sound you hear on the monitor bus. Note that all selected PFL channels will be monitored, so you can press as many PFL buttons as you like. PFL also takes over the mixer's VU meters. PFL is useful when setting the initial input gain of a channel, as it reflects the pre-fade level. PFL vs Solo PFL is similar to the solo button. There are two differences: PFL is pre-fader, solo is post-fader (i.e. the fader affects the solo level). PFL does not affect the master output but soloing a channel may do so (depending on the mixer). Sound Mixers: Channel Faders Each channel has it's own fader (slider) to adjust the volume of the channel's signal before it is sent to the next stage (subgroup or master mix). A slider is a potentiometer, or variable resistor. This is a simple control which varies the amount of resistance and therefore the signal level. If you are able to look into the inside of your console you will see exactly how simple a fader is. As a rule it is desirable to run the fader around the 0dB mark for optimum sound quality, although this will obviously vary a lot. Remember that there are two ways to adjust a channel's level: The input gain and the output fader. Make sure the input gain provides a strong signal level to the channel without clipping and leave it at that level — use the fader for ongoing adjustments. Sound Mixers: Subgroups Subgroups are a way to "pre-mix" a number of channels on a sound console before sending them to the master output mix. In the following diagram, channels 1 and 2 are assigned directly to the master output bus. Channels 3,4,5 and 6 are assigned to subgroup 1, which in turn is assigned to the master output. � Subgroups have many uses and advantages, the most obvious being that you can pre-mix (sub-mix) groups of inputs. For example, if you have six backing vocalists you can set up a good mix just for them, balancing each voice to get a nice overall effect. If you then send all six channels to one subgroup, you can adjust all backing vocals with a single subgroup slider while still maintaining the balance between the individual voices. Note that if your mixing console's subgroups are mono, you will need to use them in pairs to maintain a stereo effect. For each pair, one subgroup is the left channel and the other is right. Each channel can be panned across the two subgroups, while the subgroups are panned completely left and right into the master output bus. Sound Mixers: Outputs The main output from most mixing devices is a stereo output, using two output sockets which should be fairly obvious and easy to locate. The connectors are usually 3-pin XLRs on larger consoles, but can also be 6.5mm TR (jack) sockets or RCA sockets. The level of the output signal is monitored on the mixer's VU meters. The ideal is for the level to peak at around 0dB or just below. However you should note that the dB scale is relative and 0dB on one mixer may not be the same as 0dB on another mixer or audio device. For this reason it is important to understand how each device in the audio chain is referenced, otherwise you may find that your output signal is unexpectedly high or low when it reaches the next point in the chain. In professional circles, the nominal level of 0dB is considered to be +4 dBu. Consumer-level equipment tends to use -10 dBV. The best way to check the levels of different equipment is to use audio test tone. Send 0dB tone from the desk and measure it at the next point in the chain. Many mixers include a number of additional outputs, for example: Monitor Feed: A dedicated monitor feed which can be adjusted independently of the master output. Headphones: The headphone output may be the same as the monitor feed, or you may be able to select separate sources to listen to. Auxiliary Sends: The output(s) of the mixer's auxiliary channels. Subgroup Outputs: Some consoles have the option to output each subgroup independently. Communication Channels: Some consoles have additional output channels available for communicating with the stage, recording booths, etc. Sound Mixers: Outputs The main output from most mixing devices is a stereo output, using two output sockets which should be fairly obvious and easy to locate. The connectors are usually 3-pin XLRs on larger consoles, but can also be 6.5mm TR (jack) sockets or RCA sockets. The level of the output signal is monitored on the mixer's VU meters. The ideal is for the level to peak at around 0dB or just below. However you should note that the dB scale is relative and 0dB on one mixer may not be the same as 0dB on another mixer or audio device. For this reason it is important to understand how each device in the audio chain is referenced, otherwise you may find that your output signal is unexpectedly high or low when it reaches the next point in the chain. In professional circles, the nominal level of 0dB is considered to be +4 dBu. Consumer-level equipment tends to use -10 dBV. The best way to check the levels of different equipment is to use audio test tone. Send 0dB tone from the desk and measure it at the next point in the chain. Many mixers include a number of additional outputs, for example: Monitor Feed: A dedicated monitor feed which can be adjusted independently of the master output. Headphones: The headphone output may be the same as the monitor feed, or you may be able to select separate sources to listen to. Auxiliary Sends: The output(s) of the mixer's auxiliary channels. Subgroup Outputs: Some consoles have the option to output each subgroup independently. Communication Channels: Some consoles have additional output channels available for communicating with the stage, recording booths, etc. (Xin xem tiếp Phần 5: Chất lượng Âm thanh, Sound Quality). | |
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