Critical Listening Skills for Audio Professionals [F. Alton Everest] on bestthing.info these skills by using the intense lessons found in this book and on the CD. download Critical Listening Skills for Audio Professionals on bestthing.info ✓ FREE Story time just got better with Prime Book Box, a subscription that delivers. 5 CRITICAL LISTENING SKILLS FOR AUDIO PROFESSIONALS Talk Tech This is Hz: AMP L I T UDE 20 50 1k 2k 5k 10k 20k F RE Q U E.
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Item 54 - Critical Listening Skills for Audio Professionals developed these skills by using the intense lessons found in this book and on the CD. "Critical Listening Skills for Audio Professionals" is a book/CD combination designed to get your ears in shape so you're more effective in the. This book presents some ideas for developing critical listening skills and that the average person cannot? l How do audio professionals hear and consistently . All of the software modules are included on the accompanying CD-ROM, and .
Can each sound source be heard throughout piece? Is there any sound source that is overpowering others? Overall balance. Does the balance of musical instruments and other sound sources make sense for the music? Or is there too much of one component and not enough of another?
Is any signal level too high, causing distortion? Extraneous noise. Is there a buzz or hum from a bad cable or connection or ground problem? Technical ear training is a type of perceptual learning focused on timbral, dynamic, and spatial attributes of sound as they relate to audio recording and production. In other words, heightened listening skills can be developed allowing an engineer to analyze and rely on auditory perceptions in a more concrete and consistent way. This is not a new idea, and through years of working with audio, recording engineers generally develop strong critical listening skills.
By increasing attention on specific types of sounds and comparing successively smaller differences between sounds, engineers can learn to differentiate among features of sounds.
When two listeners, one expert and one novice, with identical hearing ability are presented with identical audio signals, an expert listener will likely be able to identify specific features of the audio that a novice listener will not. One of the goals of pursuing this type of training is to become more adept at distinguishing and analyzing a variety of timbres. Timbre is typically defined as that characteristic of sound other than pitch or loudness, which allows a listener to distinguish two or more sounds.
Timbre is a multidimensional attribute of sound and depends on a number of physical factors such as the following: All frequencies present in a sound. The relative balance of individual frequencies or frequency ranges. Primarily the attack or onset and decay time of the overall sound, but also that of individual overtones.
A person without specific training in audio or music can easily distinguish between the sound of a trumpet and a violin even if both are playing the same pitch at the same loudness—the two instruments sound different. In the world of recorded sound, engineers are often working with much more subtle differences in timbre that are not at all obvious to a casual listener.
For instance, an engineer may be comparing the sound of two different microphone preamplifiers or two digital audio sampling rates. Technical ear training focuses on the features, characteristics, and sonic artifacts that are produced by various types of signal processing commonly used in audio engineering, such as the following: Through concentrated and focused listening, an engineer should be able to identify sonic features that can positively or negatively impact a final audio mix and know how subjective impressions of timbre relate to physical control parameters.
The ability to quickly focus on subtle details of sound and make decisions about them is the primary goal of an engineer. The process of sound recording has had a profound effect on the development of music since the middle of the twentieth century. Music has been transformed from an art form that could be heard only through live performance into one where a recorded performance can be heard over and over again via a storage medium and playback system.
Sound recordings can simply document a musical performance, or they may play a more active role in applying specific signal processing and timbral sculpting to recorded sounds. With a sound recording we are creating a virtual sound stage between our loudspeakers, in which instrumental and vocal sounds are located.
Within this virtual stage recording engineers can place each instrument and sound. With technical ear training, we are focusing not only on hearing specific features of sound but also on identifying specific sonic characteristics and types of processing that cause a characteristic to be audible.
It is one thing to be able to know that a difference exists between an equalized and nonequalized recording, but it is quite another to be able to name the specific alteration in terms of center frequency, Q, and gain. Just as experts in visual art and graphic design can identify subtle shades and hues of color by name, audio professionals should be able to do the same in the auditory domain.
Sound engineers, hardware and software designers, and developers of the latest perceptual encoders all rely on critical listening skills to help make decisions about a variety of characteristics of sound and sound processing.
Many characteristics can be measured in objective ways with test equipment and test signals such as pink noise and sine tones. Some researchers such as Geddes and Lee have pointed out that high levels of measured nonlinear distortion in a device can be less perceptible to listeners than low levels of measured distortion, depending on the nature of the distortion and the testing methods employed. The opposite can also be true, in that low levels of measured distortion can be perceived strongly by listeners.
This type of situation can be true for other audio specifications such as frequency response. Listeners may prefer a loudspeaker that does not have a flat frequency response over one that does because frequency response is only one objective measurement of the total sound produced by a loudspeaker.
In other areas of audio product design, the final tuning of software algorithms and hardware designs is often done by ear by expert listeners.
Thus, physical measurements cannot be solely relied upon, and often it is auditory perceptions that determine the final verdict on sound quality.
Professionals who work with recorded sound on a daily basis understand the need to hear subtle changes in sound. It is important to know not only how these changes came about but also ways in which to use the tools available to remedy any problematic characteristics.
One of the primary goals of this book is to facilitate isomorphic mapping of technical and engineering parameters to perceptual attributes; to assist in the linking of auditory perceptions with control of physical properties of audio signals. With audio recording technology, engineers have control over technical parameters that correspond to physical attributes of an audio signal, but often it is not clear to the novice how to map a perceived sensation to the control of objective parameters of the sound.
Without extensive experience with equalizers, these numbers will have little meaning in terms of how they affect the perceived timbre of a sound. There exists an isomorphism between audio equipment that are typically used to make a recording and the type of sound an engineer hears and wishes to obtain.
An engineer can form mental links between particular features of sound quality and specific types of signal processing or equipment.
For example, a novice audio engineer may understand what the term compression ratio means in theory, but the engineer may not know how to adjust that parameter on a compressor to effectively alter the sound or may not fully understand how sound is changed when that parameter is adjusted.
One important component of teaching audio engineering is to illustrate the mapping between engineering concepts and their respective effect on the sound being heard. To teach these concepts requires the use of audio examples and also specific training for each type of processing. Ear training is equally important as knowing the functionality of equipment on hand. If an engineer uses words such as bright or muddy to describe the quality of a sound, it is not clear exactly what physical characteristics are responsible for a particular subjective quality; it could be specific frequencies, resonances, dynamics processing, artificial reverberation, or some combination of all of these and more.
There is no label on an equalizer indicating how to affect these subjective parameters. Likewise, subjective descriptions by their nature are not always consistent from person to person or across situations. It is difficult to be precise with subjective descriptions of sound, but ambiguity can be reduced if everyone agrees on the exact meaning of the adjectives being used. Continuing with the example, an equalizer requires that a specific frequency be chosen to boost or cut, but a verbal adjective chosen to describe a sound may only give an imprecise indication that the actual frequency is in the low-, mid-, or high-frequency range.
It is critical to develop an internal map of specific frequencies to perceptual attributes of a signal, and what a boost or cut at specific frequencies sounds like. With practice it is possible to learn to estimate the frequency of a deficiency or surplus of energy in the power spectrum of an audio signal and then finetune it by ear. Through years of practice, professional audio engineers develop methods to translate between their perceived auditory sensations and the technical parameters that they can control with the equipment available to them.
They also develop a highly tuned awareness of subtle details present in sound recordings. Although there may not be a common language among recording engineers to describe specific auditory stimuli, those engineers working at a very high level have devised their own personal translation between the sound they hear and imagine, and the signal processing tools available.
Comparing audiological exams between professional and novice engineers would likely not demonstrate superior hearing abilities in the professionals from a clinical, objective standpoint. Something else is going on: A recording engineer should ideally have as much command of a recording studio and its associated signal processing capability as a professional musician has command of her instrument.
A professional violinist knows precisely when and where to place her fingers on the strings and precisely what effect each bow movement will have on the sound produced. There are an intimate knowledge and anticipation of a sound even before it is produced.
An audio engineer should have this same level of knowledge and sensitivity of sound processing and shaping before reaching for an effects processor parameter, fader position, or microphone model.
There will always be times when a unique combination of signal processing and equipment choices will not be immediately apparent, but it is highly inefficient for an engineer to be continuously guessing what the standard types of studio signal processing will sound like. By knowing ahead of time what a particular parameter change will have on the sound quality of a recorded signal, an engineer can work more efficiently and effectively.
Working at such a high level, an engineer is able to respond to sound quality very quickly, similar to the speed in which musicians respond to each other in an ensemble. An engineer has direct input and influence on the artistic outcome of any music recording in which she is involved.
Listen In 2 (with Audio CD)
By adjusting balances and shaping spectra, an engineer focuses the sonic scene for listeners, guiding them aurally to a musically satisfying experience that expresses the intentions of the musical artist. An experienced recording engineer or producer can focus her attention on details of sound that may not be apparent to an untrained listener. Often, the process of making a recording from start to finish is built on hundreds, if not thousands, of decisions about technical aspects of sound quality and timbre.
Each decision contributes to a finished project and influences other choices. These decisions encompass a wide range of options and level of subtlety but typically include: Every analog component from the microphone to the input of the recording device, as well as every stage of analog to digital conversion and re-quantization will have some effect on the timbral quality of audio.
An engineer makes decisions concerning these and other technical parameters that affect the perceived audio quality and timbre of an audio signal. It may be tempting to consider these subtle changes as insignificant, but because they are added together to form a coherent whole, the cumulative effect makes each stage critical to a finished project.
Whether it is the quality of each component of a sound system or each decision made at every stage of a recording project, the additive effect is noteworthy and substantial. Choices made early in a project which degrade sound quality cannot be reversed later in a project. Audio problems cannot be fixed in the mix and, as such, engineers must be listening intently to each and every decision about signal path and processing that is made. To use an analogy, painters use specific paint colors and brush strokes in subtle ways that combine to produce powerful finished images.
In a related way, recording engineers must be able to hear and focus on specific sonic characteristics that, when taken as a whole, combine, blend, and support one another to create more powerful, meaningful final mixtures of sounds. A recording and mixing session can occupy large amounts of time, within which hundreds of subtle and not-so-subtle adjustments can be made.
The faster an engineer can home in on any sonic characteristics that may need to be changed, the more effective a given period of time will be. The ability to make quick judgments about sound quality is paramount during recording and mixing sessions.
For example, during a recording session, valuable time can be consumed while comparing and changing microphones.
What may be the perfect equalization for an instrument in one situation may not be suitable for another. What this book attempts to do, however, is guide the reader in the development of listening skills that then assist in identifying problematic areas in sound quality.
A novice engineer may not realize when there is a problem with sound quality or may have some idea that there is a problem but may not be able to identify it specifically or know how to solve it. Highly developed critical listening skills help an engineer identify characteristics of timbre and sound quality quickly and efficiently. Within each of these categories of signal processing, numerous makes and models are available at various price ranges and levels of quality.
Most compressor models have common functionalities that give them similar general sonic characteristics, but the exact way in which they perform gain reduction varies from model to model.
Differences in the analog electronics or digital signal processing algorithms among compressors create a variety of sonic results, and each make and model will have a unique sound. Through the experience of listening, engineers learn that there are variations in sound quality between different makes and models, and they will choose a certain model because of its specific sound quality. It is common to find software plug-in versions of many analog signal processing devices.
Often the screen image of a plug-in modeling an analog device will be nearly identical to the faceplate of the device.
Sometimes, because the two devices look identical, it may be tempting to think that they also sound identical. Unfortunately, they do not always sound alike but it is possible to be fooled into thinking the sound is replicated as perfectly as well as the visual representation of the device. Usually the best option is to listen and determine by ear if the two sound as similar as they look.
There is not always a direct translation between analog electronics and the computer code that performs the equivalent digital signal processing, and there are various ways to create models of analog circuits; thus we have differences in sound quality.
Although each signal processing model has a unique sound, it is possible to transfer knowledge of one model to another and be able to use an unknown model effectively after a short period of listening. Just as pianists must adjust to each new piano that they encounter, engineers must adjust to the subtle and not-so-subtle differences between pieces of equipment that perform a given function.
Sometimes timbre is the most identifying feature of a recording. In recorded music, an engineer and producer shape sounds that are captured to best suit a musical composition. The molding of timbre has become incredibly important in recorded music, and in his book The Producer as Composer: Shaping the Sounds of Popular Music , Moorefield outlines how recording and sound processing equipment contribute to the compositional process. Timbre has become such an important factor in recorded music that it can be used to identify a song before musical tonality or melody can have time to develop sufficiently.
Popular music radio stations are known to challenge listeners by playing a short excerpt typically less than a second from a well-known recording and inviting listeners to call in and identify the song title and artist.
Such excerpts are too short to indicate the harmonic or melodic progression of the music. One effect that the recording studio has had on music is that it has helped musicians and composers create sonic landscapes that are impossible to realize acoustically. In the process of recording and mixing, an engineer can manipulate any number of parameters, depending on the complexity of a mix. Many of the parameters that are adjusted during a mix are interrelated, such that by altering one track the perception of other tracks is also influenced.
The level of each instrument can affect the entire feel or focus of a mix, and an engineer and producer may spend countless hours adjusting levels—down to increments of a quarter of a decibel—to create the right balance. As an example, a slight increase in the level of an electric bass may have a significant impact on the sound and musical feel of a kick drum or even an entire mix as a whole.
Each parameter change applied to an audio track, whether it is level gain , compression, reverberation, or equalization, can have an effect on the perception of other individual instruments and the music as a whole.
Because of this interrelation between components of a mix, an engineer may wish to make small, incremental changes and adjustments, gradually building and sculpting a mix. At this point, it is still not possible to measure all perceived audio qualities with the physical measurement tools currently available.
For example, the development of perceptual coding schemes such as MPEG-1 Layer 3, more commonly known as MP3, has required the use of expert listening panels to identify sonic artifacts and deficiencies produced by data reduction processes.
Because perceptual coding relies on psychoacoustic models to remove components of a sound recording that are deemed inaudible, the only reliable test for this type of processing is the human ear. Small panels of trained listeners are more effective than large samples of the general population because they can provide consistent judgments about sound and they can focus on the subtlest aspects of a sound recording. Listeners who have completed systematic timbral ear training are able to work with audio more productively and effectively.
Recording engineers are primarily concerned with sound reproduced over loudspeakers, but there is also benefit to analyzing acoustic sound sources, as we will discuss in Chapter 7.
Single-Channel Sound Reproduction A single channel of audio reproduced over a loudspeaker is typically called monaural or mono Fig. Even if there is more than one loudspeaker, it is still considered monaural if all loudspeakers are producing exactly the same audio signal. The earliest sound recording, reproduction, and broadcast Figure 1.
Mono sound reproduction creates some restrictions for a recording engineer, but it is often this type of system that loudspeaker manufacturers use for subjective evaluation and testing of their products.
Two-Channel Sound Reproduction Evolving from monaural systems two-channel reproduction systems, or stereo, allow sound engineers greater freedom in terms of sound source location, panning, width, and spaciousness.
Stereo is the primary configuration for sound reproduction whether using speakers or headphones. Figure 1. With modestly priced headphones relative to the price of equivalent quality loudspeakers , it is possible to achieve high-quality sound reproduction. Good-quality headphones can offer Figure 1. The main disadvantage of headphones is that they create in-head localization for mono sound sources.
That is, center-panned, mono sounds are perceived to be originating somewhere between the ears because the sound is being transmitted directly into the ears without first bending around or reflecting off the head, torso, and outer ear.
To avoid in-head localization, audio signals would need to be filtered with what are known as head-related transfer functions HRTF.
Simply put, HRTFs specify filtering because of the presence of outer ears pinnae , head, and shoulders, as well as interaural time differences and interaural amplitude differences for a given sound source location. It is also worth noting that each person has a unique HRTF based on the unique shape of the outer ear, head, and upper torso.
HRTF processing has a number of drawbacks such as a negative effect on the sound quality and spectral balance and the fact that there is no universal HRTF that works perfectly for everyone.
Before downloading headphones, the reader is encouraged to listen to as many different models as possible. By comparing the sound of different headphones using music recordings that are familiar it is possible to get a better sense of the strengths and weaknesses of each model. There is no perfect headphone, and each model will have a slightly different sound. Because not all readers are near retail stores that stock high-quality headphones, some suggestions are made here at varying price points: This model is a closed design, meaning that it blocks out a substantial amount of external or background sound.
This model is also a closed back design with a comfortable circumaural fit. There are a number of models in the Grado headphone line and all are supra-aural designs, meaning that they rest right on the ear, as opposed to being circumaural, which surround the ear. Furthermore, they are all open headphones meaning that they do not block outside sound and thus might not be appropriate for listening in environments where there is significant background noise.
Grado headphones are an excellent value for the money, especially for the lower-end models, despite the fact that they are not the most comfortable headphones available. Both of these models are open design and on the higher end of the price range for headphones. They are also circumaural in design, making them comfortable to wear. These models from Sony have become somewhat of an industry standard for studio monitoring.
Multichannel Sound Reproduction Sound reproduced over more than two loudspeakers is known as multichannel, surround, ambisonic, or more specific notations indicating numbers of channels, such as 5.
Surround audio for music-only applications has had limited popularity and is still not as popular as stereo reproduction. On the other hand, surround soundtracks for film and television are common in cinemas and are becoming more common in home systems. There are many suggestions and philosophies on the exact number and layout of loudspeakers for surround reproduction systems, but the most widely accepted configuration among audio researchers is from the International Telecommunications Union ITU , which recommends a five-channel loudspeaker layout as shown in Figure 1.
Users of the ITU-recommended configuration generally also make use of an optional subwoofer or low-frequency effects LFE channel known as the. There are also more possibilities for convincing simulation of immersion within a virtual acoustic space. Feeding the appropriate signals to the appropriate channels can create a realistic sense of spaciousness and envelopment. As Bradley and Soulodre have demonstrated, listener envelopment LEV in a concert hall, a component of spatial impression, is primarily dependent on having strong lateral reflections arriving at the listener 80 ms or more after the direct sound.
There are also some challenges with respect to sound localization for certain areas within a multichannel listening area.
Summary In this chapter we have explored active listening and its importance in recording projects as well as everyday life. By defining technical ear training, we also identified some goals toward which we are working through the book and software practice modules. We finished by giving a rough overview of the main sound reproduction systems. Next we will move on to more specific ideas and exercises focused on equalization.
An audio signal with a flat spectral balance would represent all frequencies at the same relative amplitude.
Often audio engineers describe the spectral balance of sound through equalization parameters, as the equalizer is the primary tool for altering the spectral balance of sound.
An engineer can boost or cut specific frequencies or ranges of frequencies with an equalizer to bring out lowlevel details or to compensate for unwanted resonances. In the context of sound recording and production, a flat spectral balance is more likely to mean that the entire range of frequencies in a recording of a sound source is represented appropriately for a given recording project.
Is that possible or even desirable? In classical music recording, engineers usually strive for some similarity to live performances, but in most other genres of music, engineers are creating sound images that do not exist in a live performance situation.
Sounds and timbres are created and shaped in the recording studio and digital audio workstation, making it possible to take recorded sound in many possible artistic directions. Although the equalizer is the main tool for directly altering spectral balance, almost every electronic device through which audio passes alters the spectral balance of an audio signal to a greater or lesser extent. Sometimes this alteration of frequency content is necessary and completely intentional, such as with the use of equalizers and filters.
Other times a change in the spectral balance is much more subtle or nearly imperceptible, as in that caused by different types of microphone preamplifiers. Vintage audio equipment is often sought after because of unique and pleasing alterations to the spectral balance of an audio signal.
Changes in spectral balance are sometimes caused by distortion, which results in harmonics being added to an audio signal. The ability to distinguish subtle yet critical aspects of sound quality comes through the experience of listening to various types of audio processing and forming mental links between what one hears and what parameters can be controlled in an audio signal.
In essence, experienced audio professionals are like human spectral analyzers because of their ability to identify and characterize the frequency balance of reproduced sound. Aside from the use of equalizers, spectral balance can also be altered to a certain extent through dynamics processing, which changes the amplitude envelope of a signal and, by consequence, its frequency content, and through mixing a signal with a delayed version of itself, which can produce comb filtering.
Although both of these methods influence spectral balance, we are going to focus on signal processing devices whose primary function is to alter the frequency content of a signal.
An engineer seeks the equalization and spectral balance that is best suited to whatever music is being recorded. For instance, the spectral balance appropriate for a jazz drum kit recording will likely be different from that for a rock drum recording, and an experienced recording engineer, upon listening to two such audio samples, understands and can identify specific timbral differences between them. To determine the equalization or spectral balance that best suits a given recording situation, an engineer must have well-developed listening skills with regard to frequency content and its relationship to physical parameters of equalization: Each recording situation calls for specific engineering choices, and there are rarely any general recommendations for equalization that are applicable across multiple situations.
When approaching a recording project, an engineer should be familiar with existing recordings of a similar musical genre or have some idea of the timbral goals for a project to inform the decision process during production. A novice engineer may wish to employ a real-time spectral analyzer to visualize the frequency content of an audio signal and apply equalization based on what he sees.
Professional recording and mixing engineers do not usually measure the power spectrum of a music signal but instead rely on their auditory perception of the spectral balance over the course of a piece of music. Music signals generally exhibit constant fluctuations, however large or small, in frequency and amplitude of each harmonic and overtone present.
Because of the constantly changing nature of a typical music signal, it becomes difficult to get a clear reading of the amplitude of harmonics. The situation is complicated a bit more because with any objective spectral analysis there is a trade-off between time resolution and frequency resolution. With increases in time resolution, the frequency resolution decreases while the display of frequency response updates at such a fast rate that it is difficult to see details accurately while an audio signal is being played back.
Thus, physical measures currently available are not appropriate for determining what equalization to apply to a music signal, and the auditory system must be relied upon for decisions about equalization. The difference is that they have a reference, which is often pink noise or a recording, and the analyzer compares the spectrum of the original audio signal a known, objective reference to the output of the loudspeakers.
The goal in this situation is a bit different from what it is for recording and mixing because a live sound engineer is adjusting the frequency response of a sound system so that the input reference and the system output spectral balances are as similar as possible. Typically during the process of recording an acoustic musical instrument, an engineer can have direct control over the spectral balance of recorded sound, whether a single audio track or a mix of tracks, through a number of different methods.
Aside from an equalizer, the most direct tool for altering frequency balance, there are other methods available to control the spectral balance of a recorded audio track, as well as indirect factors that influence perceived spectral balance.
In this section we discuss how engineers can directly alter the spectral balance of recorded sound, as well as ways in which spectral balance can be indirectly altered during sound reproduction. The most obviously deliberate method of shaping the spectral balance of an audio signal is accomplished with an equalizer or filter, a device specifically designed to change the amplitude of selected frequencies.
Equalizers can be used to reduce particular frequency resonances in a sound recording, since they can mask other frequency components of a recorded sound and prevent the listener from hearing the truest sound of an instrument.
Besides helping to remove problematic frequency regions, equalizers can also be used to accentuate or boost certain frequency bands to highlight characteristics of an instrument or mix. There is a significant amount of art in the use of equalization, whether for a loudspeaker system or a recording, and an engineer must rely on what is being heard to make decisions about its application.
The precise choice of frequency, gain, and Q is critical to the successful use of equalization, and the ear is the final judge of the appropriateness of an equalizer setting. Engineers often choose microphones because of their unique frequency responses and how the frequency response relates to the sound source being recorded. During a beginning of a recording session, a recording engineer and producer compare the sounds of microphones to decide which ones to use for a recording.
By listening to different microphones while musicians are performing, they can decide which microphones have the sonic characteristics that are most appropriate for a given situation.
The location of a microphone in relation to a musical instrument can have a direct and clear effect on the spectral balance of the sound picked up. Sound radiated from a musical instrument does not have the same spectral balance in all directions. As an example, sound emanating directly in front of a trumpet bell will contain a much higher level of high-frequency harmonics than sound to the side of the trumpet.
An engineer can affect the frequency response of a recorded trumpet sound by simply changing the location of a microphone relative to the instrument. In this example, having the player aim the trumpet bell slightly above or below a microphone will result in a slightly darker sound than when the trumpet is aimed directly at a microphone. Even omnidirectional microphones, which are generally considered to have the best off-axis response, have some variation in their frequency response across various angles of sound incidence.
Simply changing the angle of orientation of a microphone can alter the spectral balance of a sound source being recorded. Directional microphones—such as cardioid and bidirectional polar patterns—produce an increased level of low frequencies when placed close to a sound source, in a phenomenon known as proximity effect or bass tip-up. This effect can be used to advantage to achieve prominent low frequencies when close miking a bass drum, for instance.
Because there is no direct connection between the auditory processing center of the brain and digital audio data or analog magnetic tape, engineers need to keep in mind that audio signals are altered in the transmission path between a recorder and the brain.
Three main factors influence our perception of the spectral balance of an audio signal in our studio control room: Because engineers rely on monitors to judge the spectral balance of audio signals, the frequency and power response of monitors can indirectly alter the spectral balance of the audio signals. Listening to a recording through monitors that have a weak low-frequency response, an engineer may have a tendency to boost the low frequencies in the recorded audio signal.
It is common for engineers to check a mix on three or more different sets of monitors and headphones to form a more accurate conception of what the true spectral balance of the audio signal is. Each model of loudspeaker is going to give a slightly different impression, and by listening to a variety of monitors, engineers can find the best compromise.
Beyond the inherent frequency response of a loudspeaker, almost all active loudspeakers include built-in user-adjustable filters—such as high- and low-frequency shelving filters—that can compensate for such things as low-frequency buildup when monitors are placed close to a wall. Real-time analyzers can provide some indication of the frequency response of a loudspeaker within a room, and equalizers can be used to adjust a response until it is nearly flat.
One important point to keep in mind that unless frequency response is being measured in an anechoic chamber, the response that is presented is not purely that of the loudspeaker but will also include room resonances and reflections. As we will discuss in the next section, frequency resonances in a room are prominent in some locations and less so in others.
By measuring frequency response from different locations, we average the effect of location-dependent resonances. Groups such as the International Telecommunications Union ITU have published recommendations on listening room acoustics and characteristics. Sound originating from loudspeakers propagates into a room, reflects off objects and walls, and combines with the sound propagating directly to the listener.
Sound radiates mainly from the front of a loudspeaker especially for high frequencies, but most loudspeakers become more omnidirectional as frequency decreases.
The primarily lowfrequency sound that is radiated from the back and sides of a loudspeaker will be reflected back into the listening position by any wall that may be behind the loudspeaker.
Please label your reels and supply a set of test tones for tape machine alignment. Digital There are many variables to working digitally. These are suggestions, not necessarily rules, that can be very helpful in getting the best transfer of your mixes. Most audio workstations now are 24 or 32 bit.
If you can work at 24 or 32 bit I suggest to do so. The noise floor is noticeably better than 16 bit. This has a cumulative effect when recording and mixing. Leave headroom. Please note: Berklee Online degree students are not required to walk in Commencement in Boston. You graduate when you have met all of the following criteria: Attained at least a 2. You will not officially graduate and receive your diploma until you meet all of the eligibility requirements.
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If your mailing address changes after you have submitted your graduation application, be sure to update us at graduation online. Keep in mind that if you are walking in Commencement, you will not receive your official diploma at the ceremony. Transfer Credits Can I find out how many transfer credits I am eligible for before I apply to the degree program? If you are interested in applying to the Bachelor of Professional Studies degree program and would like an estimate of the amount of transfer credit you would receive, you can request an unofficial transfer evaluation by emailing a copy of your transcript s to the Berklee Online Transfer Team at transfer online.
Be sure to include your name, major of interest, and any additional questions you may have. You can expect to receive your assessment within business days. What should I do? The earlier you contact us with questions or concerns regarding your evaluation, the easier it will be for us to address any issues.
Therefore, it is very important when you first receive your official transfer evaluation that you review the information carefully. If none of those exclusions apply, please fill out a Transfer Credit Equivalency Re-evaluation form for the courses you wish to have reconsidered.
Sometimes, we are not able to locate specific information for a course online and we are not able to determine an equivalency, but we are always happy to review additional material which will help us make that determination. No, credits completed at Berklee or through the prior learning process do not count towards the 60 transfer credit limit.
This maximum is for credit-bearing exams and undergraduate-level coursework completed externally. What is a credit deficiency and why do I need to make up credit? Credit deficiencies are caused by transferring a course that is less than three 3 credits to fulfill a three 3 credit Berklee Online requirement. Students with a credit deficiency will be short of the minimum number of credits required to graduate once they have completed their program requirements.
In order to be eligible to graduate, you will need to make up the credits you are deficient in.Simply put, HRTFs specify filtering because of the presence of outer ears pinnae , head, and shoulders, as well as interaural time differences and interaural amplitude differences for a given sound source location.
Listen In Student Book 3 with Audio CD
Is there a buzz or hum from a bad cable or connection or ground problem? This small difference in time of arrival and near-equal amplitude of direct and reflected sound at the ears of a listener creates a change in the frequency content of the sound that is heard, due to a filtering of the sound known as comb filtering.
Listening is an active process, challenging the engineer to remain continuously aware of any subtle and not so subtle perceived characteristics, changes, and defects in an audio signal. This book and the individual contributions contained in it are protected under copyright by the Publisher other than as may be noted herein. It is one thing to be able to know that a difference exists between an equalized and nonequalized recording, but it is quite another to be able to name the specific alteration in terms of center frequency, Q, and gain.
Faculty Who teaches Berklee Online courses? When two listeners, one expert and one novice, with identical hearing ability are presented with identical audio signals, an expert listener will likely be able to identify specific features of the audio that a novice listener will not.
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