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Chapter 7

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1. Strips Track Bus Options Mixing Multi cip A Track Audio 3 INSPECTOR Figure 7 19 Console view mixing view in Cakewalk Sonar 22 Digital Sound amp Music Concepts Applications amp Science Chapter 7 last updated 7 29 2013 7 2 1 2 Inputs and Outputs The original concept behind a mixer was to take the signals from multiple sources and combine them into a single audio signal that could be sent to a recording device or to an amplification system in a performance space These so called mix down consoles would have several audio input connections but very few output connections With the advent of surround sound distributed sound reinforcement systems multitrack recorders and dedicated in ear monitors most modern mixing consoles have just as many if not more outputs than inputs allowing the operator to create many different mixes that are delivered to different destinations Consider the situation of a recording session of a small rock band You could easily have more than twenty four microphones spread out across the drums guitars vocalists etc Each microphone connects to the mixing console on a separate audio input port and is fed into an input channel on the mixing console Each channel has a set of controls that allows you to optimize and adjust the volume level and frequency response of the signal and send that signal to several output channels on the mixing console Each output chan
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3. In other words the individual sends decide how much that instrument interacts with your virtual room The individual channel will deliver the dry sound to the mix and the reverb bus will deliver the wet The amount of sound that is sent on the variable aux send determines the balance of wet to dry This strategy allows you to send many different signals into the reverb processor at different levels and therefore have a separate wet dry balance for each signal while using only one reverberation processor The overall wet mix can also be easily adjusted using the fader on the aux reverb bus channel This technique is illustrated in Figure 7 28 40 Digital Sound amp Music Concepts Applications amp Science Chapter 7 last updated 7 29 2013 eoo Snowmine Mixer Arrange t M Edit Options View Single Arrange An k lt Setting Setting Setting Inserts Inserts Inserts Inserts Inserts Inserts inserts inserts Sends Sends 1 0 1 0 Jie 1 0 1 0 Jie 1 0 1 0 input5 Audio 41 Audio 44 Audio 45 Audio 48 Audio 49 Audio 50 Audio 5 Bus 8 e 53 mi o Figure 7 28 Routing each channel through a single reverb bus The third strategy for applying reverberation is to simply apply a single reverb process to an entire mix output This technique is usually not preferred because you have no control over the reverb balance between the different sounds in the mix The reaso
4. The first step in addressing some of these problems is to start using more than one microphone Stereo is the most common recording and playback technique Stereo is an entirely man made effect but produces a more dynamic effect upon playback of the recorded material with only one additional loudspeaker The basic idea is that since we have two ears two loudspeakers should be sufficient to reproduce some of the four dimensional effects of acoustic sound It s important to understand that there is no such thing as stereo sound in an acoustic space You can t make a stereo recording of a natural sound When recording sound that will be played back in stereo the most common strategy is recording each sound source with a dedicated microphone that is as acoustically isolated as possible from the other sound sources and microphones For example if you were trying to record a simple rock band you would put a microphone on each drum in the drum kit as close to the drum as possible For the electric bass you would put a microphone as close as possible to the amplifier and probably use a hardwired cable from the instrument itself This gives you two signals to work with for that instrument You would do the same for the guitar If possible you might even isolate the bass amplifier and the guitar amplifier inside acoustically sealed boxes or rooms to keep their sound from bleeding into the other microphones The singer would also be isolated in a separate room
5. is fed directly through the internal signal chain for that input channel When you connect a cable to the insert output the signal is almost always automatically rerouted away from the channel strip You ll need to feed something back into the insert input in order to continue using that channel strip on the mixing console There are two different connection designs for inserts on a mixing console The ideal design is to have a separate 14 or XLR connection for both the insert output and input This allows you to use standard patch cables to connect the external processing equipment and may also employ a balanced audio signal If the company making the mixing console needs to save space or cut down on the cost of the console they might decide to integrate both the insert output and input on a single 1 4 TRS connector In this case the input and output are handled as unbalanced signals using the tip for one signal the ring for the other signal and a shared neutral on the sleeve There is no standard for whether the input or output is carried on the tip vs the ring To use this kind of insert requires a special cable This cable has three connectors On one end is a 4 TRS connector This connector has two cables coming out of the end One cable feeds an XLR male or a 4 TS connector for the insert output and a XLR female or a 4 TS connector for the insert input 7 2 1 7 Equalizer Section After the gain section of the channel strip the next
6. it is perceived as though it s entirely located at the right loudspeaker Likewise a sound arriving 270 off axis sounds as though it s located entirely at the left loudspeaker At 0 the sound arrives at both microphones at the same level Because the sound is at an equal level in both microphones and therefore is played back equally loud through both loudspeakers it sounds to the listener as if it s coming from the phantom center image of the stereo field At 45 the polar plots tell us that the sound arrives at the right microphone approximately 7 dB louder than at the left Since this is within the 10 dB range for perception the level in the left channel causes the stereo image of the sound to be pulled slightly over from the right channel now seeming to come from somewhere between the right speaker and the phantom center location If the microphones are placed appropriately relative to the sound being recorded this technique can provide a fairly effective stereo image without requiring any 47 Digital Sound amp Music Concepts Applications amp Science Chapter 7 last updated 7 29 2013 additional mixing or panning 180 Figure 7 32 Polar patterns for two cardioid microphones in an XY cross pair Another technique for recording a live sound for a stereo effect is called mid side Mid side also uses two microphones but unlike XY one microphone is a cardioid microphone and the other is a bidirectional or figure
7. preference for the listener These effects could be as subtle as reducing an octave band of frequencies around 500 Hz by 3 dB to achieve more intelligibility for the human voice by 38 Digital Sound amp Music Concepts Applications amp Science Chapter 7 last updated 7 29 2013 allowing the higher frequencies to be more prominent The effect could be as dramatic as using a band pass filter to mimic the effect of a small cheap loudspeaker in a speakerphone When using an EQ as an effect keep in mind another rule of thumb When using an EQ you should reduce the frequencies that are too loud instead of increasing the frequencies that are too quiet Every sound system whether in a recording studio or a live performance has an amplitude ceiling the point at which the system clips and distorts If you ve done your job right you will be running the sound system at an optimal gain and a 3 dB boost of a given frequency on an EQ could be enough to cause a clipped signal Reducing frequencies is always safer than boosting them since reducing them will not blow the gain structure in your signal path 7 2 3 Applying Reverb Almost every audio project you do will likely benefit from some reverb processing In a practical sense most of the isolation strategies we use when recording sounds will have a side effect of stripping the sound of natural reverberation So anything recorded in a controlled environment such as a recording studio will probably
8. 1 ratio associated with the 55 dBFS threshold indicates that for any input signal below 55 dBFS the difference between the signal and 55 dBFS should be reduced to 1 3 the original amount When either threshold is passed 35 or 55 dBFS the attack time given on a separate panel not shown determines how long the compressor takes to achieve its target attenuation or boost When the input signal moves back between the values of 35 dBFS and 55 dBFS the release time determines how long it takes for the processor to stop applying the compression 15 Digital Sound amp Music Concepts Applications amp Science Chapter 7 last updated 7 29 2013 Graphic Tradition Settings Q ns on uo ww o A o amp hm 0 O IN ae ao 4 O d n Gn eoo 5 B 5 1 1 6o Co 2 16 Digital Sound amp Music Concepts Applications amp Science Chapter 7 last updated 7 29 2013 Figure 7 12 Dynamics processing in Adobe Audition downward and upward compression A simpler compressor one of the ARDOUR LADSPA plug ins is shown in Figure 7 13 In addition to attack release threshold and ratio controls this compressor has knee radius and makeup gain settings The knee radius allows you to shape the attack of the compression to something other than linear giving a potentially smoother transition when it kicks in The makeup gain setting often called simply gain
9. The audience makes their own sound in applause conversation shuffling in seats cell phones going off etc These sounds arrive from different directions as well Our ability to perceive this four dimensional effect is the result of the physical characteristics of our hearing system With two ears the differences in arrival time and intensity between them allow us to perceive sounds coming from many different directions Capturing this effect with audio equipment and then either reinforcing the live audio or recreating the effect upon playback is quite challenging The biggest obstacle is the microphone A traditional microphone records the sound pressure amplitude at a single point in space All the various sound waves arriving from different directions at different times are merged into a single electrical voltage wave on a wire With all the data merged into a single audio signal much of the four dimensional acoustic information is lost because when you play that recorded sound out of a loudspeaker all the reproduced sounds 44 Digital Sound amp Music Concepts Applications amp Science Chapter 7 last updated 7 29 2013 are now coming from a single direction as well Adding more loudspeakers doesn t solve the problem because then you just have every sound repeated identically from every direction and the precedence effect will simply kick in and tell our brain that the sound is only coming from the lone source that hits our ears first
10. a sound designer or composer tries to put in some underscore music or background sounds into a scene for a play or a 42 Digital Sound amp Music Concepts Applications amp Science Chapter 7 last updated 7 29 2013 film and the director inevitably says turn it down it s too loud You turn it down by 6 dB or so and the director still thinks it s too loud By the time you turn it down enough to satisfy the director you can hardly hear the sound and before long you ll be told to simply cut it because it isn t contributing to the scene in any meaningful way The secret to solving this problem is often compression When the director says the sound is too loud what he really means is that the sound is too interesting More interesting than the actor in fact and consequently the audience is more likely to pay attention to the music or the background sound than they are to the actor One common culprit when a sound is distracting is that it is too dynamic If the music is constantly jumping up and down in level it will draw your focus Using a compressor to create a less dynamic sound will often allow you to find a comfortable level for the underscore music or background sounds that will allow them to sit in the mix and enhance the scene without distracting from the performance of the actor Compression can be a useful tool but like any good thing overuse of compression can be detrimental to the quality of your sound Dynamics are
11. adverse effects on signal to noise ratio One way to think about it is that the preamplifier is where the science happens the fader is where the art happens The fader can reduce the signal level all the way to nothing or inf but typically has only five to ten dB on the amplification end of the level adjustment scale When the fader is set to 0 dB also referred to as unity the audio signal passes through with no change in level You should set the fader level to whatever sounds best and don t be afraid to move it around as the levels change over time 33 Digital Sound amp Music Concepts Applications amp Science Chapter 7 last updated 7 29 2013 E a i i o i Oo Figure 7 25 Fader and routing section of an input channel strip Near the fader there is usually a set of signal routing buttons These buttons route the audio signal at a fixed level relative to the fader position to various output channels on the 34 Digital Sound amp Music Concepts Applications amp Science Chapter 7 last updated 7 29 2013 mixing console There is almost always a main left and right stereo output labeled MIX in Figure 7 25 and sometimes a mono or center output Additionally you may also be able to route the signal to one or more group outputs or subgroup mixes A subgroup sometimes as with auxiliaries also called a bus represents a mixing channel where input signals can be grouped together under a master volum
12. again just arithmetic But even though volume changes and mixing involve simple mathematical operations they are among the most important processes we apply to audio because they potentially are very destructive Add too much to a signal and you have clipping seriously distorted audio Subtract too much and you have silence No application of filters or fancy digital signal processing can fix clipping or complete loss of signal An important form of amplitude processing is normalization which entails increasing the amplitude of the entire signal by a uniform proportion Normalizers achieve this by allowing you to specify the maximum level you want for the signal in percentages or dB and increasing all of the samples amplitudes by an identical proportion such that the loudest existing sample is adjusted up or down to the desired level This 1s helpful in maximizing the use of available bits in your audio signal as well as matching amplitude levels across different sounds Keep in mind that this will increase the level of everything in your audio signal including the noise floor r Normalize V Normalize to Decibels Format Normalize L Fi Equally DCBias Adjust 9 Help Figure 7 1 Normalizer from Adobe Audition 7 1 2 Equalization The previous section dealt with amplitude processing We now turn to processing that affects frequencies Digital Sound amp Music Concepts Applications amp Science Chapter
13. an IIR filter IIR filters also have the advantage of having analog equivalents which facilitates their design An advantage of FIR filters is that they can be constrained to have linear phase response which means that phase shifts for frequency components are proportional to the frequencies This is good for filtering music because harmonic frequencies are shifted by the same proportions preserving their harmonic relationship Another advantage of FIR filters is that they re not as sensitive to the noise that results from low bit depth and round off error 7 3 2 Low Pass High Pass Bandpass and Bandstop Filters You may have notice that in our discussion of frequency domain and time domain filters we didn t mention how we got the filters we just had them and applied them In the case of an FIR filter the filter is represented in the vector h In the case of the IIR filter the filter resides in vectors a and b Without descending the whole way through the mathematics of filter creation which is a big subject in itself we can show you algorithms for creating low pass high pass bandpass and bandstop filters when they are given the appropriate parameters as input Low pass filters allow only frequencies below a cutoff frequency f to pass through Thus Algorithm 7 2 takes f as input and outputs an N element array constituting a low pass filter Similarly Algorithm 7 3 takes f as input and yields a high pass filter and Algorithm 7 4 an
14. analog mixing console 35 Digital Sound amp Music Concepts Applications amp Science Chapter 7 last updated 7 29 2013 Also near the fader you usually have a mute button The mute button mimics the behavior of pulling the input fader all the way down In this case pre fade auxiliaries would continue to function The mute button comes in handy when you want to stop hearing a particular signal in the main mix but you don t want to lose the level you have set on the fader or lose any auxiliary functionality like signal being sent a headphone or monitor mix Instead of a mute button you may see an on off button This button shuts down the entire channel strip In that situation all signals stop on the channel including groups auxiliaries and direct outs Just to confuse you manufacturers may use the terms mute and on off interchangeably so in some cases a mute button may behave like an on off button and vice versa Check the user manual for the mixing console to find out the exact function of your button Next to the fader there is typically be a pre fade listen PFL or a solo button Pressing the PFL button routes the signal in that channel strip to a set of headphones or studio monitor outputs Since it is pre fade you can hear the signal in your headphones even if the fader is down or the mute button is pressed This is useful when you want to preview the sound on that channel before you allow it to be heard via your main or grou
15. high shelf filter to boost all those high frequencies This should be your last resort Instead you might notice that the singer is singing into the side of the microphone instead of the front Because microphones are more directional at high frequencies than low frequencies singing into the side of the microphone Max Demo 37 Digital Sound amp Music Concepts Applications amp Science Chapter 7 last updated 7 29 2013 would mean that the microphone picks up the low frequency content very easily but the high frequencies are not being captured very well In this case you would be using an EQ to boost something that isn t being picked up very well in the first place You will get much better results by simply rotating the microphone so it is pointed directly at the singer so the singer is singing into the part of the microphone that is more sensitive to high frequencies Another situation you may encounter would be when mixing the sound from multiple microphones either for a live performance or a recording You notice as you start mixing everything together that a certain instrument has a huge dip around 250 Hz You might be tempted to use an EQ to increase 250 Hz The important thing to keep in mind here is that most microphones are able to pick up 250 Hz quite well from every direction and it is unlikely that the instrument itself is somehow not generating the frequencies in the 250 Hz range while still generating all the other freque
16. inversion happening in your cables If one end of your cable is accidentally wired up incorrectly it happens more often than you might think you could have a polarity inversion when you use that cable You could take the time to re solder that connector which you should ultimately take care of but if time is short or the cable is hard to get to you could simply press the polarity button on the mixing console and instantly solve the problem There could be artistic reasons you would want to press the polarity button Consider the situation where you are trying to capture the sound of a drum If you put the microphone over the top of the drum when the drum is hit the diaphragm of the microphone pulls down towards the drum When this signal passes through your mixing console on to your loudspeakers the loudspeaker driver also pulls back away from you Wouldn t it make more sense for the loudspeaker driver to jump out towards you when the drum is hit To solve this problem you could go back to the drummer and move the microphone so it sits underneath the drum or you could save yourself the trip and just press the polarity button The audible difference here might be subtle but when you put enough subtle differences together you can often get a significant difference in audio quality Another control commonly found in the gain section is the phantom power button Phantom power is a 48 volt electrical signal that is sent down the shield of the mic
17. need some reverb added to make it sound more natural There are varying opinions on this among audio professionals Some argue that artificial reverberation processers are sounding quite good now and since it is impossible to remove natural reverberation from a recording it makes more sense to capture your recorded audio as dry as possible This way you re able to artificially add back whatever reverberation you need in a way that you can control Others argue that having musicians perform in an acoustically dry and isolated environment will negatively impact the quality of their performance Think about how much more confident you feel when singing in the shower All that reverberation from the tiled surfaces in the shower create a natural reverberation that makes your voice sound better to you than normal That gives you the confidence to sing in a way that you probably don t in public So some recording engineers would prefer to have some natural reverberation in the recording room to help the musicians to deliver a better performance If that natural reverberation is well controlled acoustically you could even end up with a recording that sounds pretty good already and might require minimal additional processing Regardless of the amount of reverb you already have in your recording you will likely still want to add some artificial reverb to the mix There are three places you can apply the reverb in your signal chain You can set it u
18. section your audio signal encounters is the equalizer section EQ shown in Figure 7 23 The number of controls you see in this section of the channel strip varies greatly across the various models of mixing consoles Very basic consoles may not include an EQ section at all Generally speaking the more money you pay for the console the more knobs and buttons you find in the EQ section We discussed the equalization process in depth in Chapter 7 29 Digital Sound amp Music Concepts Applications amp Science Chapter 7 last updated 7 29 2013 Figure 7 23 EQ section of an input channel strip Even the simplest of mixing consoles typically has two channels of EQ in each channel strip These are usually a high shelf and a low shelf filter These simple EQ sections consist of two knobs One controls the gain for the high shelf and the other for the low shelf The shelving frequency is a fixed value If you pay a little more for your mixing console you can get a third filter a mid frequency peak notch filter Again the single knob isa gain knob with a fixed center frequency and bandwidth The next controllable parameter you ll get with a nicer console is a frequency knob Sometimes only the mid frequency notch filter gets the extra variable center frequency knob but the high and low shelf filters may get a variable filter frequency using a second knob as well With this additional control you now have a semi parametric filter If you a
19. the behavior of an aux bus between pre fader and post fader while in other consoles this behavior may be fixed Sometimes this switch is located next to the aux master volume control and changes the pre fader or post fader mode for all of the channel aux sends that feed into that bus More expensive consoles allow you to select pre or post fader behavior in a channel specific way In other words each individual aux send dial on an input channel strip has its own pre or post fade button With this flexibility Aux can be set as a pre fade aux for input channel 1 and a post fade aux for input channel 2 7 2 1 9 Fader and Routing Section The fader and routing section shown in Figure 7 25 is where you usually spend most of your time working with the console in an iterative fashion during the artistic process of mixing The fader is a vertical slider control that adjusts the level of the audio signal sent to the various mixes you ve routed on that channel There are two common fader lengths 60 mm and 100 mm The 100 mm faders give your fingers greater range and control and are easier to work with The fader is primarily an attenuator It reduces the level of the signal on the channel Once you ve set the optimal level for the incoming signal with the preamplifier you use the fader to reduce that level to something that fits well in the mix with the other sounds The fader is a very low noise circuit so you can really set it to any level without having
20. the pickup patterns of the microphones overlap Figure 7 30 shows a polar plot for a cardioid microphone Recall that a cardioid microphone is a directional microphone that picks up the sound very well on axis with the front of the microphone but doesn t pick up the sound as well off axis This 45 Digital Sound amp Music Concepts Applications amp Science Chapter 7 last updated 7 29 2013 polar plot shows only one plotted line representing the pickup pattern for a specific frequency usually 1 kHz but keep in mind that the directivity of the microphone changes slightly for different frequencies Lower frequencies are less directional and higher frequencies are more directional than what is shown in Figure 7 30 With that in mind we can examine the plot for this frequency to get an idea of how the microphone responds to sounds from varying directions Our reference level is taken at 0 directly on axis The dark black line representing the relative pickup level of the microphone intersects with the 0 dB line at 0 As you move off axis the sensitivity of the microphone changes At around 75 the line intersects with the 5 dB point on the graph meaning that at that angle the microphone picks up the sound 5 dB quieter than it does on axis As you move to around 120 the microphone now picks up the sound 15 dB quieter than the on axis level At 180 the level is null o 90 180 Figure 7 30 Polar pattern for a c
21. when the sound is played through loudspeakers The inter aural isolation provided by headphones is required when listening to binaural recordings in order to get the full effect 52 Digital Sound amp Music Concepts Applications amp Science Chapter 7 last updated 7 29 2013 Figure 7 37 A binaural recording dummy head with built in microphones A few algorithms have been developed that mimic the binaural localization effect These algorithms have been implemented into binaural panning plug ins that are available for use in many DAW software programs These plug ins let you artificially create binaural effects without requiring the dummy head recordings An example of a binaural panning plug in is shown in Figure 7 38 One algorithm is called the Cetera Tutorial algorithm and is owned by the Starkey hearing aid company They use the algorithm in their hearing aids to help the reinforced sound from a hearing aid sound more like the natural response of the ear Starkey created a demo of their algorithm called the Starkey Virtual Barbershop Although this recording sounds like it was captured with a binaural recording system the Y Flash binaural localization effects are actually rendered on a computer using the Cetera algorithm 53 Digital Sound amp Music Concepts Applications amp Science Chapter 7 last updated 7 29 2013 Audio 1 Binaural Panner Figure 7 38 The binaural panning interface in Logic 7 3 Scie
22. 2 will be filtered out f samp sampling frequency of the audio signal to be filtered in Hz N the order of the filter assume N is odd Output h a bandstop FIR filter in the form of an N element array Normalize f c ando c so that mx is equal to the Nyquist angular frequency flc f1 f samp f2 c 2 f samp ol c 2 n flic 02 c 2 n f2 c middle N 2 Integer division dropping remainder for i N 2 to N 2 if i 0 h middle 1 2 f2 c f1 c else h i middle sin ol c i m i sin o2 c i n i Now apply a windowing function to taper the edges of the filter e g Hamming Hanning or Blackman Algorithm 7 5 Bandstop filter As an exercise you can try implementing these algorithms in C Java or MATLAB and see if they actually work In Section Error Reference source not found we ll show you ome higher level tools in MATLAB s digital signal processing toolkit that create these types of filters for you 7 3 3 The Convolution Theorem When the data is represented in the frequency domain it can be multiplied by a filter also in the frequency domain and certain frequencies are thereby removed or attenuated depending on the design of the filter This process is diagrammed in Figure 6 56 59 Digital Sound amp Music Concepts Applications amp Science Chapter 7 last updated 7 29 2013 audio in time domain Fourier transform audio in frequency domain mul
23. 450 400r 350 H 300 250 0 0 5 1 1 5 2 2 5 x 10 Figure 7 45 FFT of swept sine wave Store the swept sine to a PCM or uncompressed WAV audio file so you can play it back in the acoustical space you want to capture Then take your swept sine and whatever playback system you have to this space Keep in mind that the IR will inherently capture the response of your playback system as well so if you re using a cheap boombox your IR will also impart a cheap boombox response to any convolved signal Play the swept sine wave out of your loudspeaker while simultaneously recording the sound in the room Try to position your mic in a reverberant area of the room that sounds good to your ears and don t face it or the loudspeaker directly at each other Remember that you re trying to capture the sound of the room not the equipment Once you have the captured the swept sine output you re ready to obtain your IR Feel free to try out a number of captures with different mic and loudspeaker positions for your swept sine output Using MATLAB deconvolve the output recording with the original swept sine audio file MATLAB has a function for this deconv Save the result as a PCM audio file also as shown in Program 7 2 function out getIR sweptWave recordedWave sr sweptWave filename of swept sine wave recordedWave filename of sound recorded when th swept wave was played in an acoustical space SES sa
24. 7 last updated 7 29 2013 Audio equalization more commonly referred to as EQ is the process of altering the frequency response of an audio signal The purpose of equalization is to increase or decrease the amplitude of chosen frequency components in the signal This is achieved by applying an audio filter EQ can be applied in a variety of situations and for a variety of reasons Sometimes the frequencies of the original audio signal may have been affected by the physical response of the microphones or loudspeakers and the audio engineer wishes to adjust for these factors Other times the listener or audio engineer might want to boost the low end for a certain effect even out the frequencies of the instruments adjust frequencies of a particular instrument to change its timbre to name just a few of the many possible reasons for applying EQ Equalization can be achieved by either hardware or software Two commonly used types of equalization tools are graphic and parametric EQs Within these EQ devices low pass high pass bandpass bandstop low shelf high shelf and peak notch filters can be applied 7 1 3 Graphic EQ A graphic equalizer is one of the most basic types of EQ It consists of a number of fixed individual frequency bands spread out across the audible spectrum whose amplitudes can simply be turned up or down To match our non linear perception of sound the center frequencies of the bands are spaced logarithmically A graphic
25. EQ is shown in Figure 7 2 This equalizer has 31 frequency bands with center frequencies at 20 Hz 25 Hz 31 Hz 40 Hz 50 Hz 63 Hz 80 Hz and so forth in a logarithmic progression up to 20 kHz Each of these bands can be raised or lowered in amplitude individually to achieve an overall EQ shape While graphic equalizers are fairly simple to understand they are not very efficient to use since they often require that you manipulate several controls to accomplish a single EQ effect In an analog graphic EQ each slider represents a separate filter circuit that also introduces noise and manipulates phase independently of the other filters These problems have given graphic equalizers a reputation for being noisy and rather messy in their phase response The interface for a graphic EQ can also be misleading because it gives the impression that you re being more precise in your frequency processing than you actually are That single slider for 1000 Hz can affect anywhere from one third of an octave to a full octave of frequencies around the center frequency itself and consequently each actual filter overlaps neighboring ones in the range of frequencies it affects In the digital world a graphic EQ can be designed to avoid some of these problems by having the graphical sliders simply act as a user interface when in fact the slider settings are used by the DSP to build a single coherent filter Even with this enhancement graphic EQs are generally not prefer
26. Strip Show insert Y Bypass lt gt Untitled tpt WAVES TUNE Default Settings E gt Reference Shift Formant Range 0 01 14 000 0 01 16 000 0 01 18 000 I I z Receive ADI g l ozo p e ux m pe ux Mis em Edit esr Selection Apply 100 Waves Tune Lite m Figure 7 10 An autotune processor 7 1 9 Dynamics Processing 7 1 9 1 Dynamics Compression and Expansion Dynamics processing refers to any kind of processing that alters the dynamic range of an audio signal whether by compressing or expanding it As explained in Chapter 5 the dynamic range is a measurement of the perceived difference between the loudest and quietest parts of an audio signal In the case of an audio signal digitized in n bits per sample the maximum possible dynamic range is computed as the logarithm of the ratio between the loudest and the n 1 quietest measurable samples that is 20l0g4o 57 We saw in Chapter 5 that we can estimate the dynamic range as 6n dB For example the maximum possible dynamic range of a 16 bit audio signal is about 96 dB while that of an 8 bit audio signal is about 48 dB Compression 13 Digital Sound amp Music Concepts Applications amp Science Chapter 7 last updated 7 29 2013 The value of 20l0g10 dB gives you an upper limit on the dynamic range of a digital audio signal but a particular signal may
27. You can get an intuitive understanding of why the window has to be relatively small The CU purpose of the Fourier transform is to determine the frequency components of a Programming segment of sound Frequency components relate to pitches that we hear In most Exercise sounds these pitches change over time so the frequency components change over time If you do the Fourier transform on say five seconds of audio you ll have a slurring of the frequency components over time called time slurring However what if you choose a very small window size say just one sample You couldn t possibly determine any frequencies in one sample which at a sampling rate of 44 1 kHz is just 1 44 100 second Frequencies are determined by how a sound wave s amplitude goes up and down as time passes so some time must pass for there to be such a thing as frequency The upshot of this observation is that the discrete Fourier transform has to be applied over windows of samples where the windows are neither too large nor too small Note that the window size has a direct relationship with the number of frequency components you detect If your window has a size of N then you get an output telling you the magnitudes of N 2 frequency bands from the discrete Fourier transform ranging in frequency from 0 to 4 the sampling rate An overly small window in the Fourier transform gives you very high time resolution but tells you the magnitudes of only a small number of discrete wid
28. allows you to boost the entire output signal after all other processing has been applied 17 Digital Sound amp Music Concepts Applications amp Science Chapter 7 last updated 7 29 2013 GoldenMeadows SC1 by Steve Harris Presets gt Saw Figure 7 13 SC1 Compressor plug in for Ardour 7 1 9 2 Limiting and Gating A limiter is a tool that prevents the amplitude of a signal from going over a given level Limiters are often applied on the master bus usually post fader Figure 7 14 shows the LADSPA Fast Lookahead Limiter plug in The input gain control allows you to increase the input signal before it is checked by the limiter This limiter looks ahead in the input signal to determine if it is about to go above the limit in which case the signal is attenuated by the amount necessary to bring it back within the limit The lookahead allows the attenuation to happen almost instantly and thus there is no attack time The release time indicates how long it takes to go back to 0 attenuation when limiting the current signal amplitude is no longer necessary You can watch this work in real time by looking at the attenuation slider on the right which bounces up and down as the limiting is put into effect GoldenMeadows Fast Lookahead limiter by Steve Harris Figure 7 14 Limiter LADSPA plug in A gate allows an input signal to pass through only if it is above a certain threshold A hard gate has only a threshold settin
29. ardioid microphone One strategy for recording sound with a stereo effect is to use an XY cross pair The technique works by taking two matched cardioid microphones and positioning them so the microphone capsules line up horizontally at 45 angles that cross over the on axis point of the opposite microphone Getting the capsules to line up horizontally is very important because you want the sound from every direction to arrive at both microphones at the same time and therefore in the same phase 46 Digital Sound amp Music Concepts Applications amp Science Chapter 7 last updated 7 29 2013 Figure 7 31 A portable recording device using integrated microphones in an XY cross pair Figure 7 32 shows the polar patterns of both microphones when used in this configuration The signals of these two microphones are recorded onto separate tracks and then routed to separate loudspeakers for playback The stereo effect happens when these two signals combine in the air from the loudspeakers Let s first examine the audio signals that are unique to the left and right channels For a sound that arrives at the microphones 90 off axis there is approximately a 15 dB difference in level for that sound captured between the two microphones As a rule of thumb whenever you have a level difference that is 10 dB or greater between two similar sounds the louder sound takes precedence Consequently when that sound is played back through the two loudspeakers
30. at its output The center image is essentially derived from the on axis response of the mid microphone which by design happens also to be the off axis point of the side microphone Any sound that arrives at 0 to the mid microphone is added to both the left and right channels without any interaction from the signal from the side microphone since at 0 to the mid side setup the side microphone pickup is null If you look at the polar plot you can see that the mid microphone picks up every sound within a 120 spread with only 6 dB or so of variation in level Aside from this slight level difference the mid microphone doesn t contain any information that can alone be used to determine a sound s placement in the stereo field However approaching the 300 point arriving more from the left of the mid side setup you can see that the sound arriving at the mid microphone is also picked up by the side microphone at the same level and the same polarity Similarly a sound that arrives at 60 also arrives at the side microphone at the same level as the mid but this time it is inverted in polarity from the signal at the mid microphone If you look at how these two signals combine you can see that the mid sound at 300 mixes together with the Side track and because it is the same polarity it reinforces in level That same sound mixes together with the Side track 50 Digital Sound amp Music Concepts Applications amp Science Chapter 7 las
31. attenuation is already built in to the signal chain of the 1 4 input These are all factors to consider when deciding how to connect your equipment to the mixing console Another button you ll commonly find next to the gain knob is labeled This is probably the most misunderstood button in the world of sound Unfortunately the mixing console manufacturers contribute to the confusion by labeling this button with the universal symbol for phase In reality this button has nothing to do with phase This is a polarity button Pressing this button will simply invert the polarity of your signal The badly chosen symbol for the polarity button is inherited from the general confusion among sound practitioners about the difference between phase and polarity It s true that for pure sine waves a 180 degree phase shift is essentially identical to a polarity inversion But that s the only case where these two concepts intersect In the real world of sound pure sine waves are hardly ever encountered For complex sounds that you will deal with in practice phase and polarity are fundamentally different Phase changes in complex sounds are typically the result of an offset in time The phase changes as a result of timing offsets are not consistent across the frequency spectrum A shift in time that would create a 180 degree phase offset for 1 kHz would create a 360 degree phase offset for 2 kHz This inconsistent phase shift across the frequency spectrum for comp
32. bility that caused additional phase shifts and frequency combing and thus they created a more complex sound This fact hasn t stopped clever 10 Digital Sound amp Music Concepts Applications amp Science Chapter 7 last updated 7 29 2013 software developers however The flange processor shown in Figure 7 8 from Waves is one that includes a tape emulation mode and includes presets that emulate several kinds of vintage tape decks and other analog equipment eoo MetaFlanger s Untitled Presets v M Enabled Setup A Delay Tape Waveform Tupe Freq on off zi e MetaFlanger Figure 7 8 A digital flange processor 7 1 7 Vocoders A vocoder voice encoder is a device that was originally developed for low bandwidth transmission of voice messages but is now used for special voice effects in music production The original idea behind the vocoder was to encode the essence of the human voice by extracting just the most basic elements the consonant sounds made by the vocal chords and the vowel sounds made by the modulating effect of the mouth The consonants serve as the carrier signal and the vowels also called formants serve as the modulator signal By focusing on the most important elements of speech necessary for understanding the vocoder encoded speech efficiently yielding a low bandwidth for transmission The resulting voice heard at the other end of the transmission didn t have the complex frequenc
33. d Algorithm 7 5 take f and f as input to yield bandpass and bandstop filters These algorithms yield time domain filters shaped like the one in Figure 6 57 If you re interested in how these algorithms were derived see Ifeachor and Jervis 1993 Steiglitz 1996 or Burg 2008 algorithm FIR low pass filter Input f c the cutoff frequency for the low pass filter in Hz f samp sampling frequency of the audio signal to be filtered in Hz N the order of the filter assume N is odd Output h a low pass FIR filter in the form of an N element array Normalize f c and c so that mx is equal to the Nyquist angular frequency fc f c f samp oc 2 m f c middle N 2 Integer division dropping remainder for i N 2 to N 2 if i 0 h middle h middle 2 f c else h i middle sin o c i m i Now apply a windowing function to taper the edges of the filter e g Hamming Hanning or Blackman Algorithm 7 2 Low pass filter 57 Digital Sound amp Music Concepts Applications amp Science Chapter 7 last updated 7 29 2013 algorithm FIR high pass filter Input f c the cutoff frequency for the high pass filter in Hz f samp sampling frequency of the audio signal to be filtered in Hz N the order of the filter assume N is odd Output h a high pass FIR filter in the form of an N element array Normalize f c and c so that m is equal to the Nyquist angular frequenc
34. e information about the right pitch the singer needs to use with her voice In this situation you would connect her headphones to a cable that is fed from an auxiliary output which we ll call Aux 1 on the mixing console You might dial in a bit of sound to Aux 1 across each input channel of the mixing console but on the channels containing the guitar and the singer s own vocals the Aux 1 controls would be set to a higher value so they re louder in the mix being sent to the singer s headphones 21 Digital Sound amp Music Concepts Applications amp Science Chapter 7 last updated 7 29 2013 Figure 7 24 Auxiliary section of input channel strip The auxiliary send knobs on an input channel strip come in two configurations Pre Fader aux sends send signal level into the aux bus independently of the position of the channel fader In our example of the singer in the band a pre fade aux would be desirable because once you ve dialed in an aux mix that works for the singer you don t want that mix changing every time you adjust the channel fader When you adjust the channel fader it s in response to the main mix that is heard in the control room which has no bearing on what the singer needs to hear The other configuration for an aux send is Post Fader In this case dialing in the level on the aux send knob represents a level relative to the fader position for that input channel So when the main mix is changed via the fader the leve
35. e circuits and potentially collecting some noise along the way Bypassing the EQ allows you to avoid that unnecessary noise 7 2 1 8 Auxiliaries The Auxiliary controls in the channel strip are shown in Figure 7 24 Each auxiliary send knob represents an additional physical audio path output on the mixing console As you increase the value of an auxiliary send knob you re setting a certain level of that channel s signal to be sent into that auxiliary bus As each channel is added into the bus to some degree a mix of those sounds is created and sent to a physical audio output connected to that bus You can liken the function of the auxiliary busses to an actual bus transportation system Each bus or bus line travels to a unique destination and the send knob controls how much of that signal is getting on the bus to go there In most cases the mixing console will also have a master volume control to further adjust the combined signal for each auxiliary output This master control can be a fader or a knob and is typically located in the central control section of the mixing console An auxiliary is typically used whenever you need to send a unique mix of the various audio signals in the console to a specific device or person For example when you record a band the lead singer wears headphones to hear the rest of the band as well as her own voice Perhaps the guitar is the most important instrument for the singer to hear because the guitar contains th
36. e control before being passed on to the main stereo or mono output as shown in Figure 7 26 An example of subgroup routing would be to route all the drum microphones to a subgroup so you can mix the overall level of the drums in the main mix using only one fader A group is essentially the same thing except it also has a dedicated physical output channel on the mixing console The terms bus group and subgroup are often used interchangeably Group busses are almost always post fader and unlike auxiliary busses don t have variable sends it s all or nothing Group routing buttons are often linked in stereo pairs where you can use the pan knob to pan the signal between the paired groups in addition to panning between the main stereo left and right bus f f g X f 7 Y 7 e 4 Y e K Roa h A h d h Ma f h xe SOLO CLR AUX 1 AUX 2 AUX 3 AUX 4 AUX 5 AUX 6 AUX 7 AUXS MASTER MASTER nt MASTER nt MASTER 1 MASTER 1 MASTER MASTER 1 LN d AUX 1 AUX2 fAUX3 gaux4 JAuxs fPAUX6 aux auxe ER PAN 19 4 PAN 194 PAN 194 PAN 194 PAN 19 1 PAN 19 4 PAN 194 PAN 19 4 T Jj Ey 2 2 D 2 2 R A E M D A Y x 3 Ay Na Soth Ay SrA ENI Ac 2 3 Di T GRP 1 GRP 2 GRP 3 GRP 4 GRP 5 GRP 6 GRP 7 GRP 8 Figure 7 26 Master control section of an
37. e rest of the console The preamplifier is often designed for high quality and very low noise so that it can boost the audio signal without adding a lot of 25 Digital Sound amp Music Concepts Applications amp Science Chapter 7 last updated 7 29 2013 noise or distortion Because of the sheer number of electrical circuits an audio signal can pass through in a mixing console the signal can pick up a lot of noise as it travels around in the console The best way to minimize the effects of this noise is to increase the signal to noise ratio from the very start Since the preamplifier is able to increase the level of the incoming audio signal without increasing the noise level in the console you can use the preamplifier to increase the ratio between the noise floor of the mixing console and the level of your audio signal Therefore the goal of the gain knob is to achieve the highest value possible without clipping the signal Figure 7 22 Gain section of an input channel strip This is the only place in the console and likely your entire sound system where you can increase the level of the signal without also increasing the noise Thus you should get all the gain you can at this stage You can always turn the level down later in the signal chain Don t succumb to the temptation to turn down the mixing console preamplifier as a convenient way to fix problems caused downstream by power amplifiers and loudspeakers that are too powerful or
38. e right channel pan right with the polarity inverted Figure 7 34 shows a mid side matrix setup in Logic The Gain plugin inserted on the Side track is being used only to invert the polarity erroneously labeled Phase Invert in the plug in interface 49 Digital Sound amp Music Concepts Applications amp Science Chapter 7 last updated 7 29 2013 000 F Mid SideDemo Arrange co lo ps g So dm E amp E go o CBs Inspector Preferences Settings Auto Zoom Automation Flex Set Locators Repeat Section Crop Split by Playhead Bounce Regions Bounce Colors Notes Lists Media G1 amp H Etdt Track Region MIDI Audio v View v Snap Smart i Drag Overlap i e gt Global Tracks tial Mid P um a ThiinderC kO2_Sid mum PY side m m me aZ lan x M Edit v Options v View v Single Arrange All Audio Inst Aux Bus Input Output Master MIDL k 4 Setting Inserts x mm Side View Show Channel Strip Show Insert gt Bypass default Phase Invert Audio3 Stereo Out Lj Mixer Sample Editor Piano Roll Score Hyper Editor a A 01 00 00 23 16 T1151 1 120 0000 4 4 No In v a Qu e 1 2 4 103 Sei 1 130 M6 No Out Figure 7 34 Mid side matrix in Logic E Through the constructive and destructive combinations of the mid and side signals at varying angles this matrix creates a stereo effect
39. ear and perceive sounds from all directions So instead of using complicated setups with multiple microphones just by putting two microphones inside the ears of a real human you can capture exactly what the two ears are hearing This method of capture inherently includes all of the complex inter aural time and intensity difference information caused by the physical location of the ears and the human head that allows the brain to decode and perceive the direction of the sound If this recorded sound is then played back through headphones the listener perceives the sound almost exactly as it was perceived by the listener in the original recording While wearable headphone style binaural microphone setups exist sticking small microphones inside the ears of a real human is not always practical and an acceptable compromise is to use a binaural dummy head microphone A dummy head microphone is essentially the plastic head of a mannequin with molds of a real human ear on either side of the head Inside each of these prosthetic ears is a small microphone the two together capturing a binaural recording Figure 7 37 shows a commercially available dummy head microphone from Neumann With binaural recording the results are quite effective All the level phase and frequency response information of the sound arriving at both ears individually that allows us to perceive sound is maintained in the recording The real limitation here is that the effect is largely lost
40. eight microphone The cardioid microphone is called the mid microphone and is pointed forward on axis and the figure eight microphone is called the side microphone and is pointed perpendicular to the mid microphone Figure 7 33 shows the polar patterns of these two microphones in a mid side configuration 48 Digital Sound amp Music Concepts Applications amp Science Chapter 7 last updated 7 29 2013 Figure 7 33 Polar patterns for two microphones in a mid side setup The side microphone has a single diaphragm that responds to pressure changes on either side of the microphone The important thing to understand here is that because of the single diaphragm the sounds on either side of the microphone are captured in opposite polarity That is a sound that causes a positive impulse on the right of the microphone causes a negative impulse on the left of the microphone It is this polarity effect of the figure eight microphone that allows the mid side technique to work After you ve recorded the signal from these two microphones onto separate channels you have to set up a mid side matrix decoder in your mixing console or DAW software in order to create the stereo mix To create a mid side matrix you take the audio from the mid microphone and route it to both left and right output channels pan center The audio from the side microphone gets split two ways First it gets sent to the left channel pan left Then it gets sent also to th
41. el or high impedance instrument signals though you can t use both at the same time In most cases both connectors feed into the same input circuitry allowing you to use the XLR connector for line level signals as well as microphone signals This is often desirable and whenever possible you should use the XLR connector rather than the 1 4 because of its benefits such as a locking connection In some cases the 14 connector feeds into the channel strip on a separate path from the XLR connector bypassing the microphone preamplifier or encountering a 20 dB attenuation before entering the preamplifier In this situation running a line level signal through the XLR connector may result in a clipped signal because there is no gain adjustment to compensate for the increased voltage level of the line level signal Each mixing console implements these connectors differently so you ll need to read the manual to find out the specific configuration and input specifications for your mixing console 1 OQ DIRECT 0 P HI Z INPUT Figure 7 21 Input connectors for a single channel on a mixing console 7 2 1 5 Gain Section The gain section of the channel strip includes several controls The most important is the gain knob Sometimes labeled trim this knob controls the preamplifier for the input channel The preamplifier is an electrical circuit that can amplify the incoming audio signal to the optimal line level voltage suitable for use within th
42. ely separated frequencies An overly large window yields many frequencies but with poor time resolution that leads to slurring You want to have good enough time resolution to be able to reconstruct the resulting audio signal but also enough frequency information to apply the filters with proper effect Choosing the right window size is a balancing act Another interesting programming exercise is implementation of a pitch glide A Risset pitch glide is an audio illusion that sounds like a constantly rising pitch It is the aural equivalent of the visual image of a stripe on a barber pole that seems to be rising constantly Implementing the pitch glide is suggested as an exercise for this section 67 Digital Sound amp Music Concepts Applications amp Science Chapter 7 last updated 7 29 2013 C MATLAB Programming Exercise Exercise 7 3 9 Filtering and Special Effects in C 7 3 9 1 Real Time vs Off Line Processing 7 3 9 2 Dynamics Processing 7 3 10 Flange 7 4 References Flanagan J L and R M Golden 1966 Phase Vocoder Bell System Technical Journal 45 1493 1509 Ifeachor Emmanual C and Barrie W Jervis Digital Signal Processsing A Practical Approach Addison Wesley Publishing 1993 68
43. en the instruments they are being used for aren t actually playing Suppose you are operating a sound reinforcement system for a live performance and you start getting feedback through the sound system When you hear that single frequency start it s endless loop through the system you might be tempted to use an EQ to pull that frequency out of the mix This will certainly stop the feedback but all you really get is the ability to turn the system up another decibel or so before another frequency will inevitably start to feed back Repeat the process a few times and in no time at all you will have completely obliterated the frequency response of your sound system You won t have feedback but the entire system will sound horrible A better strategy for solving this problem would be to get the microphone closer to the performer and move the performer and the microphone farther away from the loudspeakers You ll get more gain this way and you can maintain the frequency response of your system We could examine many more examples of an inappropriate use of an EQ but they all go back to the rule of thumb regarding the use of an EQ as a problem solver In most cases an EQ is a very ineffective problem solver It is however a very effective tool for shaping the tonal quality of a sound This is an artistic effect that has little to do with problems of a given sound recording or reinforcement system Instead you are using the EQ to satisfy a certain tonal
44. f the filter and two vectors corresponding to the shape of the filter s frequency response Thus we can use the same f and m as before fir2 returns the vector h constituting the filter h fir2 N f m We need to use a higher order filter because this is an FIR N 30 is probably high enough The exercise associated with this section has you try MATLAB s filters for yourself 7 3 6 The Digital Signal Processing Toolkit in MATLAB 7 3 7 Creating Your Own Convolution Reverb Applying a convolution reverb to an audio signal is pretty straightforward As we saw in the previous sections all you need is an audio signal and an impulse response IR signal Then you convolve those together either using a convolution function directly on the signals Alternatively you can transform the signal and the IR into the frequency domain using an FFT and multiply their responses together transforming thee output back into a playable time domain signal Before you can apply any convolutions you ll need to locate an impulse response to use Better yet you can create your own There are a number of ways to create and capture an impulse response If you re testing an electronic system such as a piece of audio hardware or software you can simply send a short electronic pulse through it and capture what comes out the other side You can for example capture the impulse response of your sound card by generating a short pulse in MATLAB Audacity or the like a
45. f the mid microphone relative to the side microphone causes more sounds to be mixed into the left and right channels equally thereby narrowing the stereo image Unlike the XY cross pair with mid side the stereo image can be easily manipulated after the recording has already been made 2 180 Figure 7 35 Polar patterns for two microphones in mid side setup with the mid microphone attenuated 10 dB wide mode The concept behind mid side recording can be expanded in a number of ways to allow recordings to capture sound in many directions while still maintaining the ability to recreate the desired directional information on playback One example is shown in Figure 7 36 This microphone from the Soundfield Company has four microphone capsules in a tetrahedral 51 Digital Sound amp Music Concepts Applications amp Science Chapter 7 last updated 7 29 2013 arrangement each pointing a different direction Using proprietary matrix processing the four audio signals captured from this microphone can be combined to generate a mono stereo mid side four channel surround five channel surround or even a seven channel surround signal SOUNDFIELD Figure 7 36 Soundfield microphone The most simplistic and arguably the most effective method for capturing four dimensional sound is binaural recording It s quite phenomenal that despite having only two transducers in our hearing system our ears we are somehow able to h
46. filter in MATLAB is by means of the function yulewalk Let s try a low pass filter as a simple example Figure 7 43 shows the idealized frequency response of a low pass filter The x axis represents normalized frequencies and f c is the cutoff frequency This particular filter allows frequencies that are up to 4 the sampling rate to pass through but filters out all the rest fc Figure 7 43 Frequency response of an ideal low pass filter The first step in creating this filter is to store its shape This information is stored in a pair of parallel vectors which we ll call fand m For the four points on the graph in Figure 7 44 f stores the frequencies and m stores the corresponding magnitudes That is f f1 f2 f3 f 4 and m m1 m2 m3 m4 as illustrated in the figure For the example filter we have f m 0 0 25 0 25 1 1 1 0 0 Figure 7 44 Points corresponding to input parameters in yulewalk function Digital Sound amp Music Concepts Applications amp Science Chapter 7 last updated 7 29 2013 Now that you have an ideal response you use the yu ewalk function in MATLAB to determine what coefficients to use to approximate the ideal response a b yulewalk N f m Again an order V 6 filter is sufficient for the low pass filter You can use the same filter function as above to apply the filter The finite counterpart to the yulewalk function is the fir2 function Like butter fir2 takes as input the order o
47. g typically a level in dB above or below which the effect is engaged Other gates allow you to set an attack hold and release time to affect the opening holding and closing of the gate Figure 7 16 Gates are sometimes used for drums or other 18 Digital Sound amp Music Concepts Applications amp Science Chapter 7 last updated 7 29 2013 instruments to make their attacks appear sharper and reduce the bleed from other instruments unintentionally captured in that audio signal Figure 7 15 Gate Logic Pro A noise gate is a specially designed gate that is intended to reduce the extraneous noise in a signal If the noise floor is estimated to be say 80 dBFS then a threshold can be set such that anything quieter than this level will be blocked out effectively transmitted as silence A hysteresis control on a noise gate indicates that there is a threshold difference between opening and closing the gate In the noise gate in Figure 7 16 the threshold of 50 dB and the hysteresis setting of 3 dB indicate that the gate closes at 50 dBFS and opens again at 47 dBFS The side chain controls allow some signal other than the main input signal to determine when the input signal is gated The side chain signal could cause the gate to close based on the amplitudes of only the high frequencies high cut or low frequencies low cut In a practical sense there is no real difference between a gate and a noise gate A common misconcepti
48. hat reason could you possibly have to push it There are many situations where you might run into a polarity problem with one of your input signals The most common is the dreaded pin 3 hot problem In Chapter 1 we talked about the pinout for an XLR connector We said that pin 2 carries the positive or hot signal and pin 3 carries the negative or cold signal This is a standard from the Audio Engineering Society that was ratified in 1982 Prior to that each manufacturer did things differently Some used pin 2 as hot and some used pin 3 as hot This isn t really a problem until you start mixing and matching equipment from different manufacturers Let s assume your microphone uses pin 2 as hot but your mixing console uses pin 3 as hot In that situation the polarity of the signal coming into the mixing console is inverted Now if you connect another microphone to a second channel on your mixing console and that microphone also uses pin 3 as hot you have two signals in your mixing console that are running in opposite polarity In these situations having a polarity button on each channel strip is an easy way to solve this problem Despite the pin 2 hot standard being now thirty years old there are still some manufacturers making pin 3 hot equipment 27 Digital Sound amp Music Concepts Applications amp Science Chapter 7 last updated 7 29 2013 Even if all your equipment is running pin 2 hot you could still have a polarity
49. icadssaedsaleaawiansetiasanden 31 7 2 1 9 Fader and Routing SeEction cccccesseeseeeeseeeneeeeeeeenseees 33 7 2 2 Applying EQ icusietserquschoos di theo ur tz sae cux ver PR Ie NEP M P RES 37 7 2 3 Applying REVS adiscsedssixes xebussd UN nsa Ed EDMOND Exe P EN dE E 39 7 2 4 Applying Dynamics Processing eeseeeeeenn mnn 42 7 2 5 Applying Spettal Effects oiii bie pa qu bien prese ubdukbst tubo usns 43 45 Creating Stereo isst uoa Xp RERUUEDERKRKEOGU EU QU PEU dp OUR bI RU Ra 44 7 2 7 Capturing the Four Dimensional Sound Field 44 7 3 Science Mathematics and Algorithms cccceeee essen eens eeeeeenaeee 54 7 3 1 Convolution and Time Domain Filtering ceeeeeeeesse 54 7 3 2 Low Pass High Pass Bandpass and Bandstop Filters 57 7 3 3 The Convolution Theorem eeeeeeeeeenm nnn 59 7 3 4 Diagramming Filters and Delays eee 61 7 3 5 FIR and IIR Filters in MATLAB suicscsesacus cua e RyRA VY x ExEe Eug med 61 7 3 6 The Digital Signal Processing Toolkit in MATLAB 63 7 3 7 Creating Your Own Convolution Reverb seen 63 7 3 8 Experiments with Filtering Vocoders and Pitch Glides 66 7 3 9 Filtering and Special Effects in C sssssssssnrrrrnsnrrrrrsnrrnne 68 7 3 9 1 Real Time vs Off Line Processing seeee 68 29 9 2 Dynamics PrOCESSING oas
50. ime domain convolve convolution filter in time domain Time Domain Filtering equivalent result Fourier transform filter in frequency domain multiply inverse Fourier transform audio in frequency domain Frequency Domain Filtering Figure 7 42 The Convolution Theorem 7 3 4 Diagramming Filters and Delays 7 3 5 FIR and IIR Filters in MATLAB The previous section gives you algorithms for creating a variety of FIR filters MATLAB also provides functions for creating FIR and IIR filters Let s look at the IIR filters first MATLAB s butter function creates an IIR filter called a Butterworth filter named for its creator The butter function call a b butter N f sends in two arguments the order of the desired filter N and the and the cutoff frequency f It should be noted that the cutoff frequency is normalized so that the Nyquist frequency 7 the sampling rate is 1 and all valid frequencies lie between 0 and 1 The function call returns two vectors a and b corresponding to the vectors a and b in Equation 7 3 For a simple low pass filter an order of 6 is fine Now with the filter in hand you can apply it using the fi ter function The filter function takes the coefficients and the vector of audio samples as arguments output filter a b audio 61 Digital Sound amp Music Concepts Applications amp Science Chapter 7 last updated 7 29 2013 Another way to create and apply an IIR
51. ion which is where this reverb effect gets its name Applying convolution reverb as a filter is like passing the audio signal through a representation of the original room itself This makes the audio sound as if it were propagating in the same acoustical space as the one in which the impulse response was originally captured adding its reverberant characteristics With convolution reverb processors you lose the extra control provided by the traditional pre delay early reflections and RT60 parameters but you often gain a much more natural reverberant effect Convolution reverb processors are typically more CPU intensive than their more traditional counterparts but with the speed of modern CPU s this is not a big concern Figure 7 7 shows an example of a convolution reverb plug in Digital Sound amp Music Concepts Applications amp Science Chapter 7 last updated 7 29 2013 X Audio 1 Ca View Show Channel Strip Show Insert Bypass Compare 4 OL3s Choir Hall Reset All 1 3s Choir Hall 1 314s Stereo sample rate pL E latency c Zoom to fit ajo Oo rev vol compensation Reverb Spread llolume Enuelope HP m 7 Pre Dly Spread init level attack time decay time enp lin end level C Space Designer Figure 7 7 A convolution reverb processor from Logic 7 1 6 Flange Flange is the effect of combing out frequencies in a continuously changing frequency range The flange effect i
52. is case It s helpful to think of Equation 7 1 algorithmically as described Algorithm 7 1 The notation y n indicates that the nt output sample is created from a convolution of input values from the audio signal x and the filter multipliers in h as given in the summation Keep in mind that we have to get an output value for every input so the equation is applied as many times as there are samples Thus the equation is repeated in a loop for every sample in the audio input Input X an array of digitized audio samples i e in the time domain of size M h a convolution filter of size N Specifically a finite impulse response filter FIR Output y the audio samples filtered y for n 0toN 1 y n h n Q x n Xgzo h e x k where x n k Oifn k lt 0 Algorithm 7 1 Convolution with a finite impulse response FIR filter The FIR convolution process is described diagrammatically in Figure 7 40 55 Digital Sound amp Music Concepts Applications amp Science Chapter 7 last updated 7 29 2013 audio in time domain convolution filter in time domain filtered audio convolve a gt in time domain closeup of convolution h A4 h 3 h 2 h 1 h 0 filter x 0 x 1 x 2 x 3 x 4 x 5 x 6 x 7 x 8 x 9 audio samples y n h 4 x 0 h 3 x 1 h 2 x 2 h 1 x 3 h 0 x 4 for a filter of size 5 y is the filtered audio in the ti
53. ks remain unaffected original downward expanded audio expansion dynamic range WT ALL original upward compressed audio compression dynamic range dau illi EMI m ND samples below threshold are boosted samples below threshold are reduced Figure 7 11 Dynamics compression and expansion Adobe Audition has a dynamics processor with a large amount of control Most dynamics processor s controls are simpler than this allowing only compression for example with the threshold setting applying only to downward compression Audition s processor allows settings for compression and expansion and has a graphical view and thus it s a good one to illustrate all of the dynamics possibilities Figure 7 12 shows two views of Audition s dynamics processor the graphic and the traditional with settings for downward and upward compression The two views give the same information but in a different form In the graphic view the unprocessed input signal is on the horizontal axis and the processed input signal is on the vertical axis The traditional view shows that anything above 35 dBFS should be compressed at a 2 1 ratio This means that the level of the signal above 35 dBFS should be reduced by 2 Notice that in the graphical view the slope of the portion of the line above an input value of 35 dBFS is This slope gives the same information as the 2 1 setting in the traditional view On the other hand the 3
54. l in that aux send is changed as well This is particularly useful when you re using an aux bus for some kind of effect processing In our same recording session example you might want to add some reverberation to the mix Instead of inserting a separate reverb processor on each input channel requiring multiple processors it s much simpler to connect an aux output on the mixing console to the input of a single reverb processor The output of the reverb processor then comes back into an unused input channel on the mixing console This way you can use the aux sends to dial in the desired amount of reverb for each input channel The reverb processor then returns a reverberant mix of all of the sounds that gets added into the main mix Once you get a good balance of reverb dialed in on an aux send for a particular input channel you don t want that balance to change If the aux send to the reverb is pre fader when the fader is used to adjust the channel level within the main mix the 32 Digital Sound amp Music Concepts Applications amp Science Chapter 7 last updated 7 29 2013 reverb level remains the same disrupting the balance you achieve Instead when you turn up or down the channel fader the level of the reverb should also increase or decrease respectively so the balance between the dry and the reverberant wet sound stays consistent Using a post fader aux send accomplishes this goal Some mixing consoles give you a switch to change
55. ld This reduces the dynamic range e Downward expansion attenuates signals that are below a given threshold not changing signals above the threshold This increases the dynamic range e Upward expansion boosts signals that are above a given threshold not changing signals below the threshold This increases the dynamic range The common parameters that can be set in dynamics processing are the threshold attack time and release time The threshold is an amplitude limit on the input signal that triggers compression or expansion The same threshold triggers the deactivation of compression or expansion when it is passed in the other direction The attack time is the amount of time allotted for the total amplitude increase or reduction to be achieved after compression or expansion is triggered The release time is the amount of time allotted for the dynamics processing to be turned off reaching a level where a boost or attenuation is no longer being applied to the input signal 14 Digital Sound amp Music Concepts Applications amp Science Chapter 7 last updated 7 29 2013 samples above threshold are reduced samples above threshold are boosted Eu MEME MEL EU i downward compressed original upward expanded gudi expansion dynamic range MT ll jd Jl ull Pu ues UD m compression dynamic range F quiet parts remain unaffected quiet parts remain unaffected peaks remain unaffected pea
56. le like the one pictured in Figure 7 17 you might feel intimidated by all the knobs and buttons It s important to realize that most of the controls are simply duplicates Each input channel is represented by a vertical column or channel strip of controls as shown in Figure 7 20 It s good to realize that the audio signal typically travels through the channel strip and its various controls from top to bottom This makes it easy to visualize the audio signal path and understand how and when the audio signal is being affected For example you ll typically find the preamp gain control at the top of the channel strip as this is the first circuit the audio signal encounters while the level fader at the bottom is the last component the signal hits as it leaves the channel strip to be mixed with the rest of the individual signals 23 Digital Sound amp Music Concepts Applications amp Science Chapter 7 last updated 7 29 2013 F a ekRaeo Rea X j Figure 7 20 A single channel strip from an analog mixing console 24 Digital Sound amp Music Concepts Applications amp Science Chapter 7 last updated 7 29 2013 7 2 1 4 Input Connectors Each input channel has at least one input connector as shown in Figure 7 21 Typically this is an XLR connector Some mixing consoles also have a 4 TRS connector on each input channel The idea for including both is to use the XLR connector for microphone signals and the 4 connector for line lev
57. lex sounds is the cause of comb filtering when two identical sounds are mixed together with an offset in time Given that a mixing console is all about mixing sounds it is very easy to cause comb filtering when mixing two microphones that are picking up the same sound at two different distances resulting in a time offset If you think the button in question adjusts the phase of your signal as the symbol on the button suggests you might come to the conclusion that pressing this button will manipulate the timing of your signal and compensate for comb filter problems Nothing could be further from the truth In a comb filter situation pressing the polarity button for one of the two signals in question will simply convert all cancelled frequencies into frequencies that reinforce each other All the frequencies that were reinforcing each other will now cancel out Once you ve pressed this button you still have a comb filter It s just an inverted comb filter When you encounter two channels on your console that cause a comb filter when mixed together a better strategy is to simply eliminate one of the two signals After all if these two signals are identical enough to cause a comb filter you don t really need both of them in your mix do you Simply ducking the fader on one of the two channels will solve your comb filter problem much more efficiently and certainly more so than using the polarity button If this button has nothing to do with phase w
58. me domain Figure 7 40 Filtering in the time domain by convolving with an FIR filter IIR filters are also time domain filters but the process by which they work is a little different To describe an IIR we need a filter of infinite length given by this equation y n h n Q x n h k x n k where x n k 0ifn k 0 and k is theoretically infinite Equation 7 2 IIR Filter infinite form We can t deal with an infinite summation in practice but Equation 7 2 can be transformed to a difference equation form which gives us something we can work with N 1 y n h n 8 x n X ax k byn k k 0 where x n k Oifn k z 0 Equation 7 3 IIR filter difference equation form In Equation 7 3 N is the order of the forward filter and M is the order of the feedback filter The output from an IIR filter is determined by convolving the input and combining it with the feedback of previous output In contrast the output from an FIR filter is determined solely by convolving the input FIR and IIR filters each have their advantages and disadvantages In general FIR filters require more memory and processor time IIR filters can more efficiently create a sharp cutoff between frequencies that are filtered out and those that are not An FIR filter requires a larger 56 Digital Sound amp Music Concepts Applications amp Science Chapter 7 last updated 7 29 2013 filter size to accomplish the same sharp cutoff as
59. mpling rate 65 Digital Sound amp Music Concepts Applications amp Science Chapter 7 last updated 7 29 2013 swept wavread recordedWave corded wavread recordedWave deconvolve to get the impulse response which can be used as a filter on other another audio signal to make it sound as if it is being played in the original acoustical space H o o9 oe oe out r deconv recorded swept plot out soundsc out sr wavwrite out sr IR wav end Program 7 2 Using deconvolution to capture an impulse response to use as a filter In an audio editor such as Audacity you ll want to clean up the IR file a bit trimming out any extra silence before the impulse and after the impulse decays Your IR file shouldn t be any longer than the reverb time of the room you captured likely no more than several seconds long at most You could try to program some of this cleanup into your MATLAB function but sometimes it s easier to do it visually looking at the waveform If you happen to listen to your IR file at this time you ll probably notice it sounds like a sort of pop with an unusual timbre as mentioned earlier in the chapter You can now take this IR and load it into a compatible convolution reverb plugin in your DAW or use MATLAB to convolve it with an audio signal of your choice 7 3 8 Experiments with Filtering Vocoders and Pitch Glides Vocoders were introduced in Sec
60. n you would use this technique is if you don t have access to the raw tracks or if you are trying to apply a special reverb effect to a single audio file In this case just pick a reverb setting and adjust the wet dry mix until you achieve the sound you are looking for The most difficult task in using reverb is to find the right balance It is very easy to overdo the effect The sound of reverberation is so intoxicating that you have to constantly fight the urge to apply the effect more dramatically Before you commit to any reverb effect listen to it though a few different speakers or headphones and in a few different listening environments A reverb effect sounds like a good balance in one environment might sound over the top in another listening environment Listen to other mixes of similar music or sound to compare what you have done with the work of seasoned professionals Before long you ll develop a sixth sense for the kind of reverb to apply in a given situation 41 Digital Sound amp Music Concepts Applications amp Science Chapter 7 last updated 7 29 2013 7 2 4 Applying Dynamics Processing When deciding whether to use dynamics processing you should keep in mind that a dynamics processor is simply an automatic volume knob Any time you find yourself constantly adjusting the level of a sound you may want to consider using some sort of dynamics processor to handle that for you Most dynamics processors are in the form of downwards c
61. nce Mathematics and Algorithms 7 3 1 Convolution and Time Domain Filtering In earlier chapters we showed how an audio signals can be represented in either the time domain or the frequency domain In this section you ll see how mathematical operations are applied in these domains to implement filters delays reverberation etc Let s start with the time domain Filtering in the time domain is done by a convolution operation Convolution uses a convolution filter which is an array of N values that when graphed takes the basic shape shown in Figure 7 39 A convolution filter is also referred to as a convolution mask an impulse response or a convolution kernel There are two commonly used time domain convolution filters that are applied to digital audio They are FIR filters finite impulse response and IIR filters infinite impulse response 54 Digital Sound amp Music Concepts Applications amp Science Chapter 7 last updated 7 29 2013 CAT Figure 7 39 Graph of time domain convolution filter Equation 7 1 describes FIR filtering mathematically N 1 y n h n x n 2 h k x n k where x n k Oif n k lt 0 Equation 7 1 FIR filter By our convention boldface variables refer to vectors i e arrays In this equation A n is the convolution filter essentially a vector of multipliers to be applied successively to audio samples The number of multipliers is the order of a filter N in th
62. ncies reasonably well So before you turn on that EQ you should mute all the other channels on the mixer and listen to the instrument alone If the problem goes away you know that whatever is causing the problem has nothing to do with EQ In this situation comb filtering is the likely culprit There s another microphone in your mix that was nearby and happened to be picking up this same instrument at a slightly longer distance of about two feet When you mix these two microphones together 250 Hz is one of the frequencies that cancels out If that isn t the issue try moving a foot or two closer to or farther away from the loudspeakers If the 250 Hz dip goes away in this case there s likely a standing wave resonance in your studio at the mix position that is cancelling out this frequency Using an EQ in this case will not solve the problem since you re trying to boost something that is actively being cancelled out A better solution for the standing wave would be to consider rearranging your room or applying acoustical treatment to the offending surfaces that are causing this reflective build up If comb filtering was the issue you should try to better isolate the signals either by moving the microphones farther apart or preventing them from being summed together in the mix A gate might come in handy here too If you gate both signals you can minimize the times when both microphones are mixed together since the signals won t be let through wh
63. nd sending it out of your soundcard through an audio cable that goes back into an input of your soundcard to record it back into software Of course your soundcard is designed to have a very clean response so your IR won t be very interesting An interesting experiment however would be to play that pulse through an EQ plugin set to some particular filtering before sending it out of your sound card The pulse you capture on the way back in will be an IR with the response of that EQ filter If you then take that IR and convolve it with an audio signal you should get the same effect as if you sent the audio signal through the EQ itself You can even compare your convolved results with an actual equalized signal in your DAW You can try this experiment with an artificial reverb plug in as well If you want to capture the impulse response of a physical acoustical space or system you ll need a more robust method To capture a decent impulse response you need to excite all of the audible frequencies in an acoustical space There are a number of ways to do this but the most widely used and effective method is using a swept sinusoid signal A swept sine is a pure 63 Digital Sound amp Music Concepts Applications amp Science Chapter 7 last updated 7 29 2013 tone signal whose frequency varies over time typically sweeping across the audible spectrum over some specified duration We ve discussed that playing a short electronic pulse impulse
64. nd she was actually singing the note at 435 Hz the autotuner would detect the discrepancy and make the correction If you think about how an autotuner might be implemented you ll realize the complexities involved Suppose you record a singer singing just the note A which she holds for a few seconds Even if she does this nearly perfectly her voice contains not just the note A but harmonic overtones that are positive integer multiples of the fundamental frequency Your algorithm for the software autotuner first must detect the fundamental frequency call it from among all the harmonics in the singer s voice It then must determine the actual semitone nearest to f Finally it has to move f and all of its harmonics by the appropriate adjustment All of this sounds possible when a single clear note is steady and sustained long enough for your algorithm to analyze it But what if your algorithm has to deal with a constantly changing audio signal which is the nature of music Also consider the dynamic pitch modulation inherent in a singer s vibrato a commonly used vocal technique Detecting individual notes separating them one from the next and snapping each sung note and all its harmonics to appropriate semitones is no trivial task An example of an autotune processor is shown in Figure 7 10 12 Digital Sound amp Music Concepts Applications amp Science Chapter 7 last updated 7 29 2013 e Audio 1 d View Show Channel
65. nel represents a different mix of the signals from the various microphones The main mix output channel likely contains a mix of all the different microphones and is sent to a pair or more of monitor loudspeakers in the control room for the recording engineer and other participants to listen to the performance from the band This main mix may also represent the artistic arrangement of the various inputs decided upon by the engineer producer and band members eventually intended for mixed down distribution as a stereo or surround master audio file Each performer in the band is also often fed a separate auxiliary output mix into her headphones Each auxiliary mix contains a custom blend of the various instruments that each musician needs to hear in order to play his part in time and in tune with the rest of the band Ideally the actual recording is not a mix at all Instead each input channel has a direct output connection that sends the microphone signal into a dedicated channel on a multitrack recording device which in the digital age is often a dedicated computer DAW This way the raw isolated performances are captured in their original state and the artistic manipulation of the signals can accomplished incrementally and non destructively during the mixing process 7 2 1 3 Channel Strips Configuring all the knobs buttons and faders on a suitably sized mixing console makes all of the above functions possible When you see a large mixing conso
66. not occupy that full range You might have a signal that doesn t have much difference between the loudest and quietest parts like a conversation between two people speaking at about the same level On the other hand you might have at a recording of a Rachmoninoff symphony with a very wide dynamic range Or you might be preparing a background sound ambience for a live production In the final analysis you may find that you want to alter the dynamic range to better fit the purposes of the recording or live performance For example if you want the sound to be less obtrusive you may want to compress the dynamic range so that there isn t such a jarring effect from a sudden difference between a quiet and a loud part In dynamics processing the two general possibilities are compression and expansion each of which can be done in the upwards or downwards direction Figure 7 11 Generally compression attenuates the higher amplitudes and boosts the lower ones the result of which is less difference in level between the loud and quiet parts reducing the dynamic range Expansion generally boosts the high amplitudes and attenuates the lower ones resulting in an increase in dynamic range To be precise e Downward compression attenuates signals that are above a given threshold not changing signals below the threshold This reduces the dynamic range e Upward compression boosts signals that are below a given threshold not changing signals above the thresho
67. now become more common Figure 7 18 and as you can see they look pretty much the same as their analog counterparts Software mixers with user interfaces modeled after equivalent hardware are a standard part of audio processing programs like Pro Tools Apple Logic Ableton Live and Cakewalk Sonar The mixing view for a software mixer is sometimes called the console view as is the case with Cakewalk Sonar pictured in Figure 7 19 20 Digital Sound amp Music Concepts Applications amp Science Chapter 7 last updated 7 29 2013 ere eee ddl Figure 7 17 Analog mixing console 5 6 STEREO STIN1 STIN2 sE sa SEL SEL SEL CUE 73 PARAMA Be ON on ON ON ON ON ON o ER ERI Sr 9 9 0 v 0 0 o 3 32 5 moie MASTER Figure 7 18 A digital mixing console In the following section we introduce the different components and functions of mixers Whether a mixer is analog or digital hardware or software is not the point The controls and 2i Digital Sound amp Music Concepts Applications amp Science Chapter 7 last updated 7 29 2013 functions of mixers are generally the same no matter what type you re dealing with or the context in which you re doing the mixing Bg MakingMusic cwp SONAR X2 Producer oo File Edit Views Insert Process Project Utilities Window Help E MakingMusic cwp Track i Modules
68. ompressors These compressors work by reducing the level of sounds that are too loud but letting quieter sounds pass without any change in level One example when compression can be helpful is when mixing multiple sounds together from a multitrack recording The human voice singing with other instruments is typically a much more dynamic sound than the other instruments Guitars and basses for example are not known as particularly dynamic instruments A singer is constantly changing volume throughout a song This is one of the tools a singer uses to produce an interesting performance When mixing a singer along with the instruments from a band the band essentially creates a fairly stable noise floor The word noise is not used here in a negative context rather it is used to describe a sound that is different from the vocal that has the potential of masking the vocal if there is not enough difference in level between the two As a rule of thumb for adequate intelligibility of the human voice the peaks of the voice signal need to be approximately 25 dB louder than the noise floor which in this case is the band It is quite possible for a singer to perform with a 30 dB dynamic range In other words the quietest parts of the vocal performance are 30 dB quieter than the loudest parts of the vocal performance If the level of the band is more or less static and the voice is moving all around how are you going to maintain that 25 dB ratio between the peak
69. on is that noise gates can be used to remove noise in a recording In reality all they can really do is mute or reduce the level of the noise when only the noise is present Once any part of the signal exceeds the gate threshold the entire signal is allowed through the gate including the noise Still it can be very effective at clearing up the audio in between words or phrases on a vocal track or reducing the overall noise floor when you have multiple tracks with active regions but no real signal perhaps during an instrumental solo 19 Digital Sound amp Music Concepts Applications amp Science Chapter 7 last updated 7 29 2013 9077 0 Untitled3 Figure 7 16 Noise gate Logic Pro 7 2 Applications 7 2 1 Mixing 7 2 1 1 Mixing Contexts and Devices A mixing console or mixer is a device that takes several different audio signals and mixes them together typically to be sent to another device in a more consolidated or organized manner Mixing can be done in a variety of contexts Mixing during a live performance requires that an audio engineer balance the sounds from a number of sources Mixing is also done in the sound studio as the recordings from multiple channels or on multiple tracks are combined Mixing can also be done with a variety of tools An audio engineering doing the mixing of a live performance could use a hardware device like the one shown in Figure 7 17 an analog mixing console Digital mixers have
70. one quality of sound and music that makes it exciting interesting and evocative A song with dynamics that have been completely squashed will not be very interesting to listen to and can cause great fatigue on the ears If you apply compression inappropriately it may cause audible artifacts in the sound where you can noticeably hear when the sound is being attenuated and released This is often referred to as pumping or breathing and usually means you ve taken the compression to far or in the wrong direction So be very strategic about how you use compression and go easy on the compression ratio when you do use it Often a mild compression ratio is enough to tame an overly dynamic sound without completely stripping it of all its character 7 2 5 Applying Special Effects One of the most common special effects is using delay to create an echo effect This is used often in popular music The challenge with a delay effect is to synchronize the timing of the echoes with the beat of the music If you are using a delay plug in with a DAW program the plug in will try to use the metronome of your project file to create the delay timing This works if you recorded the music to the system s metronome but if you just recorded everything freestyle you will need to synchronize the delay manually Typically this is done with a tap pad Also called tap delay these plug ins use a pad or button that you can tap along with the beat of the music to keep
71. p as an insert for a specific channel in a multi channel mix In this case the reverb only gets applied to the one specific channel and the other channels are left unchanged You will have to adjust the wet dry mix in the reverb processor to create an appropriate balance This technique can be useful for a special effect you want to put on a specific sound but using this technique on every channel in a large multi channel mix will cost 39 Digital Sound amp Music Concepts Applications amp Science Chapter 7 last updated 7 29 2013 you a lot in CPU performance because of all the discrete reverb processors that are running simultaneously If you have a different reverb setting on each channel you could also have a rather confusing mix since every sound will seem to be in a different acoustic environment Maybe that s what you want if you re creating a dream sequence or something abstract for a play or film but for a music recording it usually makes more sense to have every instrument sounding like it is in the same room The second reverb technique can solve both the problem of CPU performance and varying acoustic signatures In this case you would set up a mix bus that has a reverb inserted You would set the reverb processor to 100 wet This basically becomes the sound of your virtual room Then you can set up each individual channel in your mix to have a variable aux send that dials in a certain amount of the signal into the reverb bus
72. p outputs If you have a solo button when pressed it will also mute all the other channels allowing you to hear only the solo enabled channels Solo is typically found in recording studio consoles or audio recording software Sometimes the terms PFL and solo are used interchangeably so again check the user manual for your mixing console to be sure of the function for this button Similar to PFL is after fade listen AFL AFL is typically found on output faders allowing you to preview in your headphones the signal that is passing through a subgroup group aux or main output The after fade feature is important because it allows you to hear exactly what is passing through the output including the level of the fader For example if a musician says that he can t hear a given instrument in his monitor you can use the AFL feature for the aux that feeds that monitor to see if the instrument can be heard If you can hear it in your headphones then you know that the aux is functioning properly In this case you may need to adjust the mix in that aux to allow the desired instrument to be heard more easily If you can t hear the desired instrument in your headphones then you know that you have a routing problem in the mixing console that s preventing the signal from sending out from that aux output Depending on the type of mixing console you re using you may also have some sort of PPM Peak Programme Meter near the fader In some cases this will be a
73. peak notch filter the frequency parameter corresponds to the center frequency of the band to which the filter is applied For the low pass high pass low shelf and high shelf filters which don t have an actual center the frequency parameter represents the cut off frequency The numbered circles on the frequency response curve correspond the filter bands Figure 7 4 shows a low pass filter in band 1 where the 6 dB downpoint the point at which the frequencies are attenuated by 6 dB is set to 500 Hz Digital Sound amp Music Concepts Applications amp Science Chapter 7 last updated 7 29 2013 Band Filter Freq Gain JOJ Ae 0 room A i i Figure 7 4 Low pass filter in a n EQ with cut off frequency of 500 Hz The gain parameter is the amount by which the corresponding frequency band will be boosted or attenuated The gain cannot be set for low or high pass filters as these types of filters are designed to eliminate all frequencies beyond or up to the cut off frequency The Q parameter is a measure of the height vs the width of the frequency response curve A higher Q value creates a steeper peak in the frequency response curve compared to a lower one as shown in Figure 7 5 Some parametric equalizers use a bandwidth parameter instead of Q to control the range of frequencies for a filter Bandwidth works inversely from Q in that a larger value of bandwidth represents a larger range of frequencies The unit of measu
74. re given a third knob to control the filter Q or Bandwidth the filter becomes fully parametric From there you simply get more bands of fully parametric filters per channel strip as the cost of the console increases Depending on your needs you may not require five bands of EQ per channel strip The option that is absolutely worth paying for is an EQ bypass button This button routes the audio signal in the channel around the EQ circuit This way the audio signal doesn t have to be processed by the EQ if you don t need any adjustments to the frequency response of the signal Routing around the EQ solves two potential problems The first is the problem of inheriting someone else s solution There are a lot of knobs on a mixing console and they aren t always 30 Digital Sound amp Music Concepts Applications amp Science Chapter 7 last updated 7 29 2013 reset when you start working on a new project If the EQ settings from a previous project are still dialed in you could be inheriting a frequency adjustment that s not appropriate for your project Having an EQ bypass button is a quick way to turn off all the EQ circuits so you re starting with a clean slate The bypass button can also help you quickly do an A B comparison without having to readjust all of the filter controls The second problem is related to noise floor Even if you have all the EQ gain knobs flattened out no boost or cut your signal is still passing though all thos
75. red by experiences professionals Digital Sound amp Music Concepts Applications amp Science Chapter 7 last updated 7 29 2013 100Hz 200Hz 400Hz 1000Hz 2000Hz 400 gem Draw Curves Graphic EQ Length of Filter sect ouve named s Enan iain imei ont a Gea Figure 7 2 Graphic EQ in Audacity 7 1 4 Parametric EQ A parametric equalizer as the name implies has more parameters than the graphic equalizer making it more flexible and useful for professional audio engineering Figure 7 3 shows a parametric equalizer The different icons on the filter column show the types of filters that can be applied They are from top to bottom peak notch also called bell low pass high pass low shelf and high shelf filters The available parameters vary according to the filter type This particular filter is appling a low pass filter on the 4 band and a high pass filter on the 5 band Digital Sound amp Music Concepts Applications amp Science Chapter 7 last updated 7 29 2013 E Ey Band Filter Freq Q Gain aded 50 nef 240 ooa ALES 2 2 100 ne 20 ood ALES 3 E 150 nef 20 00a 4 A 200 He so o0 ee 5 250 He sod oou EE s 300 He 240 00 MB ESSE Output 00 8 i Sonitus equalizer cakewalk Brhh tano Figure 7 3 Parametric EQ in Cakewalk Sonar For the
76. rement for bandwidth is typically an octave A bandwidth value of 1 represents a full octave of frequencies between the 6 dB down points of the filter Figure 7 5 Comparison of Q values for two peak filters Digital Sound amp Music Concepts Applications amp Science Chapter 7 last updated 7 29 2013 7 1 5 Reverb When you work with sound either live or recorded the sound is typically captured with the microphone very close to the source of the sound With the microphone very close and particularly in an acoustically treated studio with very little reflected sound it is often desired or even necessary to artificially add a reverberation effect to create a more natural sound or perhaps to give the sound a special effect Typically a very dry initial recording is preferred so that artificial reverberation can be applied more uniformly and with greater control There are several methods for adding reverberation Before the days of digital processing this was accomplished using a reverberation chamber A reverberation chamber is simply a highly reflective isolated room with very low background noise A loudspeaker is placed at one end of the room and a microphone is placed at the other end The sound is played into the loudspeaker and captured back through the microphone with all the natural reverberation added by the room This signal is then mixed back into the source signal making it sound more reverberant Reverberation chambe
77. rophone cable to power condenser microphones In our example there is a dedicated 48 volt phantom power button for each input channel strip In some consoles there s a global phantom power button that turns on phantom power for all inputs The last control that is commonly found in the gain section of the console is a high pass filter Pressing this button filters out frequencies below the cutoff frequency for the filter Sometimes this button has a fixed cutoff frequency of 80Hz 100Hz or 125Hz Some mixing consoles give you a knob along with the button that allows you to set a custom cutoff frequency for the high pass filter When working with microphones it s very easy to pick up unwanted sounds that have nothing to do with the sound you re trying to capture Footsteps pops wind and handling noise from people touching and moving the microphone are all examples of unwanted sounds that can show up in your microphone The majority of these sounds fall in very low frequencies Most musical instruments and voices do not generate frequencies below 125 Hz so you can safely use a high pass to filter out frequencies lower than that Engaging this filter removes most of these unwanted sounds before they enter the signal chain in your system without affecting the good sounds you re trying to capture Still all filters have an effect on the phase of the frequencies surrounding the cutoff frequency and they can introduce a small amount of additional noise in
78. rs vary in size and construction some larger than others but even the smallest ones would be too large for a home much less a portable studio Because of the impracticality of reverberation chambers most artificial reverberation is added to audio signals using digital hardware processors or software plug ins commonly called reverb processors Software digital reverb processors use software algorithms to add an effect that sounds like natural reverberation These are essentially delay algorithms that create copies of the audio signal that get spread out over time and with varying amplitudes and frequency responses A sound that is fed into a reverb processor will come out of that processor with thousands of copies or virtual reflections As described in Chapter 4 there are three components of a natural reverberant field A digital reverberation algorithm attempts to mimic these three components The first component of the reverberant field is the direct sound This is the sound that arrives at the listener directly from the sound source without reflecting from any surface In audio terms this is known as the dry or unprocessed sound The dry sound is simply the original unprocessed signal passed through the reverb processor The opposite of the dry sound is the wet or processed sound Most reverb processors include a wet dry mix that allows you to balance the direct and reverberant sound Removing all of the dry signal leaves you with a very ambien
79. s created by adding two identical audio signals with one slightly delayed relative to the other usually on the order of milliseconds or samples The effect involves continuous changes in the amount of delay causing the combed frequencies to sweep back and forth through the audible spectrum In the days of analog equipment like tape decks flange was created mechanically in the following manner Two identical copies of an audio signal usually music were played simultaneously and initially in sync on two separate tape decks A finger was pressed slightly against the edge called the flange of one of the tapes slowing down its rpms This delay in one of the copies of the identical waveforms being summed resulted in the combing out of a corresponding fundamental frequency and its harmonics If the pressure increased continuously the combed frequencies swept continuously through some range When the finger was removed the slowed tape would still be playing behind the other However pressing a finger against the other tape could sweep backward through the same range of combed frequencies and finally put the two tapes in sync again Artificial flange can be created through mathematical manipulation of the digital audio signal as shown in the exercise associated with Section 7 3 10 However to get a classic sounding flanger you need to do more than simply delay a copy of the audio This is because tape decks used in analog flanging had inherent varia
80. s of the voice and the level of the band In this situation you will never find a single level for the vocal fader that will allow it to be heard and understood consistently throughout the song You could painstakingly draw in a volume automation curve in your DAW software or you could use a compressor to do it for you If you can set the threshold somewhere in the middle of the dynamic range of the vocal signal and use a 2 1 or 4 1 compression ratio can easily turn that 30 dB of dynamic range into a 20 dB range or less Since the compressor is turning down all the loud parts the compressed signal will sound much quieter than the uncompressed signal but if you turn the signal up using either the output gain of the compressor or the channel fader you can bring it back to a better level With the compressed signal you can now much more easily find a level for the voice that allows it to sit well in the mix Depending on how aggressive you are about the compression you may still need to automate a few volume changes but the compressor has helped turn a very difficult to solve problem into something more manageable Rather than using a compressor to allow a sound to more easily take focus over a background sound you can also use compression as a tool for getting a sound to sit in the mix in a way that allows other sounds to take focus This technique is used often in theatre and film for background music and sound effects The common scenario is when
81. sasssus beue E RRRRREERRRRROEERERA ERR RR FERE AGE 68 Paola e cm 68 7A References cixscsexesezxk yep x DER FEX pagi ap X DU C RIEN Mx MEAE DERE R 68 This material is based on work supported by the National Science Foundation under CCLI Grant DUE 0717743 Jennifer Burg Pl Jason Romney Co PI Digital Sound amp Music Concepts Applications amp Science Chapter 7 last updated 7 29 2013 7 Chapter 7 Audio Processing 7 1 Concepts 7 1 1 Amplitude Adjustments and Mixing We ve entitled this chapter Audio Processing as if this is a separate discrete topic within the realm of sound But actually everything we do to audio is a form of processing Every tool plug in software application and piece of gear is essentially an audio processor of some sort What we set out to do in this chapter is to focus on particular kinds of audio processing covering the basic concepts applications and underlying mathematics of these One of the most straightforward types of audio processing is amplitude adjustment something as simple as turning up or down a volume control In the analog world a change of volume is achieved by changing the voltage of the audio signal In the digital world it s achieved by adding to or subtracting from the sample values in the audio stream just simple arithmetic The mixing of two digital audio signals is another simple example of audio processing Digital mixing is accomplished by adding sample values together
82. sound The reverberant sound is made of up all the remaining reflections that have bounced around many surfaces before arriving at the listener These reflections are so numerous and close together that they are perceived as a continuous sound Each time the sound reflects off a surface some of the energy is absorbed Consequently the reflected sound is quieter than the sound that arrives at the surface before being reflected Eventually all the energy is absorbed by the surfaces and the reverberation ceases Reverberation time is the length of time it takes for the reverberant sound to decay by 60 dB effectively a level so quiet it ceases to be heard This is sometimes referred to as the RT60 or also the decay time A longer decay time indicates a more reflective room Because most surfaces absorb high frequencies more efficiently than low frequencies the frequency response of natural reverberation is typically weighted toward the low frequencies In reverberation processors there is usually a parameter for reverberation dampening This applies a high shelf filter to the reverberant sound that reduces the level of the high frequencies This dampening variable can suggest to the listener the type of reflective material on the surfaces of the room Figure 7 6 shows a popular reverberation plug in The three sliders at the bottom right of the window control the balance between the direct early reflection and reverberant sound The other controls adjus
83. t effect as if the actual sound source was not in the room at all The second component of the reverberant field is the early reflections Early reflections are sounds that arrive at the listener after reflecting from the first one or two surfaces The number of early reflections and their spacing vary as a function of the size and shape of the room The early reflections are the most important factor contributing to the perception of room size In a larger room the early reflections take longer to hit a wall and travel to the listener In a reverberation processor this parameter is controlled by a pre delay variable The longer the pre delay the longer time you have between the direct sound and the reflected sound giving the effect of a larger room In addition to pre delay controls are sometimes available for determining the number of early reflections their spacing and their amplitude The spacing of the early reflections indicates the location of the listener in the room Early reflections that are spaced tightly together give the effect of a listener who is closer to a side or corner of the room The amplitude of the early reflections suggests the distance from the wall On the other hand low amplitude reflections indicate that the listener is far away from the walls of the room Digital Sound amp Music Concepts Applications amp Science Chapter 7 last updated 7 29 2013 The third component of the reverberant field is the reverberant
84. t the setting for each of these three components of the reverberant field 4 Mediurn Concert Time Response Dimension Room Size Distance Balance Decay Time Pre Delay Density Gee EN C ER Lowcut Rew Shelf ER Absorb Hi Fre 3 BW G9 Go Ez 64s Low Reverb Damping High 3609 Frequency Response VFHREEEERERUEU ET ELE EET UQUN E TrueVerb Figure 7 6 The TrueVerb reverberation plug in from Waves Digital Sound amp Music Concepts Applications amp Science Chapter 7 last updated 7 29 2013 The reverb processor pictured in Figure 7 7 is based on a complex computation of delays and filters that achieve the effects requested by its control settings Reverbs such as these are often referred to as algorithmic reverbs after their unique mathematical designs There is another type of reverb processor called a convolution reverb which creates its effect using an entirely different process A convolution reverb processor uses an impulse response IR captured from a real acoustic space such as the one shown in Figure 7 7 An impulse response is essentially the recorded capture of a sudden burst of sound as it occurs in a particular acoustical space If you were to listen to the IR which in its raw form is simply an audio file it would sound like a short pop with somewhat of a unique timbre and decay tail The impulse response is applied to an audio signal by a process known as convolut
85. t the top of the console on a meter bridge Cheaper consoles will just give you two LED indicators one for when audio signal is present and another for when the signal clips More expensive consoles will give you high resolution PPMs with several different level indicators A PPM is more commonly found in digital systems but is also used in analog equipment A PPM is typically a long column of several LED indicators in three different colors as shown in Figure 7 27 One color represents signal levels below the nominal operating level another color represents signals at or above nominal level and the third color usually red represents a signal that is clipping or very near to clipping A PPM responds very quickly to the audio signal Therefore a PPM is very useful for measuring peak values in an audio signal If you re trying to find the right position for a preamplifier a PPM will show you exactly when the signal clips Most meters in audio software are programmed to behave like a PPM 36 Digital Sound amp Music Concepts Applications amp Science Chapter 7 last updated 7 29 2013 M dB m Hardware PPM meter Software PPM meter Figure 7 27 PPM meters 7 2 2 Applying EQ An equalizer can be incredibly useful when used appropriately and incredibly dangerous when used inappropriately Knowing when to use an EQ is just as important and knowing how to use it to accomplish the effect you are looking for Every time you
86. t updated 7 29 2013 and cancels out because of the polarity inversion The sound that arrives from the left of the mid side setup therefore is louder on the left channel and accordingly appears to come from the left side of the stereo field upon playback Conversely a sound coming from the right side at 60 reinforces when mixed with the Side track but cancels out when mixed with the Side track and the matrixed result is louder in the right channel and accordingly appears to come from the right of the stereo field Sounds that arrive between 0 and 300 or 0 and 60 have a more moderate reinforcing and canceling effect and the resulting sound appears at some varying degree between left right and center depending on the specific angle This creates the perception of sound that is spread between the two channels in the stereo image The result here is quite similar to the XY cross pair technique with one significant difference Adjusting the relative level of the Mid track alters the spread of the stereo image Figure 7 35 shows a mid side polar pattern with the mid microphone attenuated 10 dB Notice that the angle where the two microphones pick up the sound at equal levels has narrowed to 45 and 315 This means that when they are mixed together in the mid side matrix a smaller range of sounds are mixed equally into both left and right channels This effectively widens the stereo image Conversely increasing the level o
87. the echoes synchronized Usually after eight taps the echoes get in sync with the music but as the performance from the musician changes you ll need to periodically re tap the plug in Figure 7 29 shows a tap delay processor with the mouse pointer on the tap pad 43 Digital Sound amp Music Concepts Applications amp Science Chapter 7 last updated 7 29 2013 a LdVoxC12 cmp 03 o View Show Channel Strip Show Insert Y Bypass Compare Ste F EQ Section On Off Gain Rotate ela On Off Tupe Frec Mode Tep Feedback Snap ie Grid Seteenths v Grid M ode erm SuperTap F SuperTap 2 Taps m gt s Figure 7 29 A tap delay plug in Other special effects including flangers pitch shifting autotune etc may be applied in several different situations There are really no rules with special effects Just make sure you have a real reason for using the effect and don t overdo it 7 2 6 Creating Stereo 7 2 7 Capturing the Four Dimensional Sound Field When listening to sound in an acoustic space such as at a live orchestral concert you hear different sounds arriving from many directions The various instruments are spread out on a stage and their sound arrives at your ears somewhat spread out in time and direction according to the physical location of the instruments You also hear subtly nuanced copies of the instrument sounds as they are reflected from the room surfaces at even more varying times and directions
88. the swept sine as well as precise measurement microphones to capture it Of course feel free to use what you have available in order to try this out Omnidirectional mics tend to work best The impulse response will still be effective to some degree The first thing you need to do is generate an appropriate swept sine signal to be used as the input to your acoustical system Make one that sweeps from 20 Hz to 20 kHz logarithmically over a 15 second period You can easily do this in MATLAB as shown in Program 7 1 The FFT of the wave is shown in Figure 7 45 function sweep start stop secs A Start start frequency Stop stop frequency secs number of seconds for the sweep A amplitude Run with sweep 20 20000 15 1 to get a one second sweep from frequencies 0 to 10 000 Hz at a sampling rate of 44 1 kHz At least 10 samples are needed for the last frequency sr stop 10 if sr gt 44100 sr 44100 end N sr secs fl start sr 64 Digital Sound amp Music Concepts Applications amp Science Chapter 7 last updated 7 29 2013 f2 stop sr a 2 pix f2 f1 N b 2 pi fl1 for i 1 N 1 sweep i A sin a power i 2 2 b i end fftdata fft sweep magFreqs abs fftdata plot magFreqs 1 sr 2 soundsc sweep sr wavwrite sweep sr sweep wav end Program 7 1 A swept sine wave in MATLAB 700 650r 600 I 550r 500r
89. think you want to use an EQ you should evaluate the situation against this rule of thumb EQ should be used to create an effect not to solve a problem Using an EQ as a problem solver can cause new problems when you should really just figure out what s causing the original problem and fix that instead Only if the problem can t be solved in any other way should you pull up the EQ perhaps if you re working post production on a recording captured earlier during a film shoot or you ve run into an acoustical issue in a space that can t be treated or physically modified Rather than solving problems you should try to use an EQ as a tool to achieve a certain kind of sound Do you like your music to be heavy on the bass An EQ can help you achieve this Do you really like to hear the shimmer of the cymbals in a drum set An EQ can help Let s examine some common problems you may encounter where you will be tempted to use an EQ inappropriately As you listen to the recording you re making of a singer you notice that the recorded audio has a lot more low frequency content than high frequency content leading to a decreased intelligibility You go over and stand next to the performer to hear what they actually sound like and notice that they sound quite different than what you are hearing from the microphone Standing next to them you can hear all those high frequencies quite well In this situation you may be tempted to pull out your EQ and insert a
90. through a system results in an impulse response Clearly a swept sine signal is not an impulse so simply passing it through a system is not going to output an IR in the same way With a little mathematics however we can retrieve the IR We know that a filtered output signal is achieved by convolving a known input signal with a filter s impulse response or multiplying them in the frequency domain In this case if we capture the output of the swept sine wave played in an acoustic environment our filter we know both the output what we captured as well as the input our swept sine signal So solving for the unknown impulse response you ll see we can obtain it by deconvolving our output by the input as expressed in Equation 7 4 In this case f is the swept sine wave g is the sound we record when we play the swept wave in a chosen acoustical space and h is the filter we seek to apply to other signals so that they sound like they are played in the same acoustical space If g f h then h g f where is the convolution operator and 1 is the deconvolution operator Equation 7 4 Note that we can also perform this operation in the frequency domain This is accomplished by dividing the output frequency response by the input frequency response and performing the inverse FFT to get back the time domain IR Typically to get the best result when capturing an IR you ll want to employ a good flat loudspeaker to play back
91. tion 7 1 7 The implementation of a vocoder is sketched in Algorithm 7 6 and diagrammed in Figure 7 46 The MATLAB and C exercises associated with this section encourage you to try your hand at the implementation algorithm vocoder Input c an array of audio samples constituting the carrier signal m n array of audio samples constituting the modulator signal Output v the carrier wave modulated with the modulator wave Initialize v with Os Divide the carrier into octave separated frequency bands with bandpass filters Divide the modulator into the same octave separated frequency bands with bandpass filters for each band use the modulator as an amplitude envelope for the carrier Algorithm 7 6 Sketch of an implementation of a vocoder 66 Digital Sound amp Music Concepts Applications amp Science Chapter 7 last updated 7 29 2013 modulator signal bandpass filters envelope followers apply amplitude envelopes band by band frequeng bands of carrier signal gt bandpass filters EE carier signal vocoded result Figure 7 46 Overview of vocoder implementation One thing to note if you try to implement the vocoder is that the Fourier transform is not applied to an entire audio signal at one time Rather it s applied in small sections called windows on the order of about 1024 samples If you use the Fast Fourier transform the window size must be a power of 2
92. tiply filter in frequency domain filtered audio in frequency domain inverse Fourier transform filtered audio in time domain Figure 7 41 Filtering in the frequency domain MATLAB has a function called fft for performing the Fourier transform on a vector of audio data However to get a closer view of these operations it may be enlightening to try implementing the Fourier transform yourself and comparing your results with the results of MATLAB s transform as suggested in the exercise Fourier theory has shown that filtering in the frequency domain can be done such that it gives results equivalent to filtering in the time domain That is if you take a time domain filter transform it to the frequency domain transform your audio data to the frequency domain multiply the frequency domain filter and the frequency domain audio and do the inverse Fourier transform on the result you ll get the same result as applying the time domain filter on the time domain audio data This is known as the convolution theorem and is explained diagrammatically in Figure 7 42 In fact with a fast implementation of the Fourier transform known as the Fast Fourier Transform FFT filtering in the frequency domain is more computationally efficient than filtering in the time domain 60 Digital Sound amp Music Concepts Applications amp Science Chapter 7 last updated 7 29 2013 audio in time domain filtered audio in t
93. to the signal For this reason you should leave the high pass filter disengaged unless you need it 7 2 1 6 Insert Next to the channel input connectors typically there is a set of insert connections Insert connections consist of an output and input that allow you to connect some kind of external processing device in line with the signal chain in the channel strip The insert output typically takes the audio signal from the channel directly after it exits the preamplifier though some consoles let you choose at what point in the signal path the insert path lies Thinking back to the top down signal flow the insert connections are essentially inserting an extra component at that point on the channel strip In this case the component isn t built into the channel strip like the EQ or pan controls Rather the device is external and can be whatever the engineer wishes 28 Digital Sound amp Music Concepts Applications amp Science Chapter 7 last updated 7 29 2013 to use If for example you want to compress that dynamics of the audio on input channel 1 you can connect the insert output from channel 1 to the input of an external compressor Then the output of the compressor can be connected to the insert input on channel 1 of the mixing console The compressed signal is then fed back into the channel strip and continues down the rest of the signal chain for channel 1 If nothing is connected to the insert ports it is bypassed and the signal
94. too sensitive for your application Also you should not turn down the preamplifier in an effort to get all the channel faders to line up in a straight row These are excellent ways to create a noisy sound system because you re decreasing the signal to noise ratio for the incoming audio signal Once you ve set that gain knob to the highest level you can without clipping the signal the only reason you should ever touch it again is if the signal coming in to the console gets louder and starts clipping the input If you re feeding a line level signal into the channel you might find that you re clipping the signal even though the gain knob is turned all the way down Most mixing consoles have a pad button next to the gain knob This pad button sometimes labeled 20 dB Line range or Mic Line will attenuate the signal by 20 dB which should allow you to find a setting on your gain knob that doesn t clip Using the pad button shouldn t necessarily be something you do automatically when using line level signals as you re essentially undoing 20 dB of built in signal to noise ratio Don t use it unless you have to Be aware that sometimes this button also serves to reroute the input signal using the 4 input instead of the XLR On some consoles that 26 Digital Sound amp Music Concepts Applications amp Science Chapter 7 last updated 7 29 2013 have both 4 and XLR inputs yet don t have a pad button it s because the 20 dB
95. with a dedicated microphone During the recording process the signal from each microphone is recorded on a separate track in the DAW software and written to a separate audio file on the hard drive With an isolated recording of each instrument a mix can be created that distributes the sound of each instrument between two channels of audio that are routed to the left and right stereo loudspeaker To the listener sitting between the two loudspeakers a sound that is found only on the left channel sounds like it comes from the left of the listener and vice versa for the right channel A sound mixed equally into both channels appears to the listener as though the sound is coming from an invisible loudspeaker directly in the middle This is called the phantom center channel By adjusting the balance between the two channels you can place sounds at various locations in the phantom image between the two loudspeakers This flexibility in mixing is possible only because each instrument was recorded in isolation This stereo mixing effect is very popular and produces acceptable results for most listeners When recording in a situation where it s not practical to use multiple microphones in isolation such as for a live performance or a location recording where you re capturing an environmental sound it s still possible to capture the sound in a way that creates a stereo like effect This is typically done using two microphones and manipulating the way
96. y fc f c f samp Qe 2 m f c middle N 2 Integer division dropping remainder for i N 2 to N 2 if i 0 h middle 1 2 f c else h i middle sin o c i n i Now apply a windowing function to taper the edges of the filter e g Hamming Hanning or Blackman Algorithm 7 3 High pass filter algorithm FIR bandpass filter Input fl the lowest frequency to be included in Hz f2 the highest frequency to be included in Hz f samp sampling frequency of the audio signal to be filtered in Hz N the order of the filter assume N is odd Output h a bandpass FIR filter in the form of an N element array Normalize f c and o c so that m is equal to the Nyquist angular frequency flc f1 f samp f2 c 2 f samp Ql c 2 n flic 02 2 n f2 c middle N 2 Integer division dropping remainder for i N 2 to N 2 if i 0 h middle 2 f2 c 2 flic else h i middle sin o2 c i m i sin wl_c i n i Now apply a windowing function to taper the edges of the filter e g Hamming Hanning or Blackman Algorithm 7 4 Bandpass filter 58 Digital Sound amp Music Concepts Applications amp Science Chapter 7 last updated 7 29 2013 algorithm FIR bandstop filter Input fl the highest frequency to be included in the bottom band in Hz f2 the lowest frequency to be included in the top band in Hz Everything from f1 to
97. y components of a real human voice but enough information was there for the words to be intelligible Today s vocoders used in popular music combine voice and instruments to make the instrument sound as if it s speaking or conversely to make a voice have a robotic or techno sound The concept is still the same however Harmonically rich instrumental music serves as the carrier and a singer s voice serves as the modulator An example of a software vocoder plug in is shown in Figure 7 9 11 Digital Sound amp Music Concepts Applications amp Science Chapter 7 last updated 7 29 2013 amp Audio 1 View Show Channel Strip Show Insert Y Side Chain None Bypass Compare lt gt Carrier Modulator Noise Morph 2271 Filter Fre Bypass 200 e 0 0 J 0 0 j 0 0 J 0 0 J 0 0 Pressure 20 0 Ff 258 J 688 1804 4476 11046 Formant 1 00 Intemal orga 1 ajro 0 190 100 090 Smoothing f 60 0 0 gt a L nfa L lt fa Release 100 BMA NEKN E1151 LIEUAIJ 11 2E 1 4E IEA LR 2223 Morphoder Morphoder s Figure 7 9 A vocoder processor 7 1 8 Autotuners An autotuner is a software or hardware processor that is able to move a pitch of the human voice to the frequency of the nearest desired semitone The original idea was that if the singer was slightly off pitch the autotuner could correct the pitch For example if the singer was supposed to be on the note A at a frequency of 440 Hz a

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