Audio Production II - Spring 2011

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Audio Production II - Spring 2011 - Quiz

Final Study Guide Quiz
Note: This quiz was created by a student to aid in the Prepration for a final exam. The answers to this test are in no form approved by any higher education as correct, but have been highly researched by the creator. Not all subjects are converet, but only the more difficult terms and questions discussed since the midterm.


Questions and Answers
  • 1. 

    Energy from the Microphone to the DAW and then back to the speakers is converted through the following process: Microphone -> Preamp (Boost to line level) -> AD Converter (Audio to Digital Signal Processing) -> DAW -> FX (within DAW usually using sends and bus) -> Playback (on Speakers)

    • A.

      True

    • B.

      False

    Correct Answer
    B. False
    Explanation
    After FX it is send to a DA Converter which allows the audio to be played on the speakers

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  • 2. 

    What is an XLR Cable and what do each of the pins do?

    • A.

      XLR connectors are normally used for transmitting balanced mic and line level signals. Pin 1 of an XLR connector is always ground/shield. The connectors are designed so that pin 1 makes its connection first when inserted in a jack; this ensures that the ground connection is made first, helping prevent pops and thumps in the audio chain. Either pin 2 or pin 3 may be hot (determined by the gear the connector is plugged into), with the remaining pin being cold. To maintain correct polarity in a signal path, it is important to be aware of which pin is hot or cold on all connections, and wire your cables accordingly.

    • B.

      XLR connectors are normally used for transmitting balanced mic and line level signals. Pin 1 of an XLR connector is always hot. The connectors are designed so that pin 2 makes its connection first when inserted in a jack; this ensures that the ground connection is made first, helping prevent pops and thumps in the audio chain. Either pin 2 or pin 3 may be ground (determined by the gear the connector is plugged into), with the remaining pin being cold. To maintain correct polarity in a signal path, it is important to be aware of which pin is hot or cold on all connections, and wire your cables accordingly.

    • C.

      XLR connectors are normally used for transmitting unbalanced mic, line and instrument level signals. Pin 1 of an XLR connector is always hot. The connectors are designed so that pin 2 makes its connection first when inserted in a jack; this ensures that the ground connection is made first, helping prevent pops and thumps in the audio chain. Either pin 2 or pin 3 may be ground (determined by the gear the connector is plugged into), with the remaining pin being cold. To maintain correct polarity in a signal path, it is important to be aware of which pin is hot or cold on all connections, and wire your cables accordingly.

    Correct Answer
    A. XLR connectors are normally used for transmitting balanced mic and line level signals. Pin 1 of an XLR connector is always ground/shield. The connectors are designed so that pin 1 makes its connection first when inserted in a jack; this ensures that the ground connection is made first, helping prevent pops and thumps in the audio chain. Either pin 2 or pin 3 may be hot (determined by the gear the connector is plugged into), with the remaining pin being cold. To maintain correct polarity in a signal path, it is important to be aware of which pin is hot or cold on all connections, and wire your cables accordingly.
    Explanation
    XLR connectors are commonly used for transmitting balanced mic and line level signals. Pin 1 is always ground/shield and makes its connection first when inserted in a jack, ensuring that the ground connection is made first to prevent audio disturbances. Either pin 2 or pin 3 can be hot, depending on the gear the connector is plugged into, while the remaining pin is cold. It is crucial to know which pin is hot or cold on all connections to maintain correct polarity and wire the cables accordingly.

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  • 3. 

    Which type of connector is a 1/4" (or 1/8") balanced connector with three sections of the shaft. The sections inclue 2 connectors plus a ground (shild) in one plug.

    • A.

      TRS - Tip, Ring, and Sleeve

    • B.

      TT - Tiny Telephone

    • C.

      TS - Tip and sleeve

    • D.

      ADAT - Alesis Digital Audio Tape

    Correct Answer
    A. TRS - Tip, Ring, and Sleeve
    Explanation
    The correct answer is TRS - Tip, Ring, and Sleeve. This type of connector is commonly used in audio devices to carry balanced signals. The tip carries the positive audio signal, the ring carries the negative audio signal, and the sleeve is the ground/shield. By using three sections, this connector helps to eliminate noise and interference, resulting in a cleaner audio signal.

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  • 4. 

    TT - Abbreviation for Tip Sleeve. Tip Sleeve refers to a specific type of phone plug (not phono plug) or 1/4" connector that is set up for a two-conductor ____________ connection. The tip and sleeve are separated by an insulator. The Tip is generally considered the "hot," or where the signal is applied, while the Sleeve is where the ground or shield is connected.

    Correct Answer
    unbalanced
    Unbalanced
    Explanation
    The correct answer is "unbalanced." An unbalanced connection refers to a type of audio connection that uses a two-conductor cable, with the tip carrying the signal and the sleeve serving as the ground or shield. This type of connection is commonly used in consumer audio devices and has a higher susceptibility to noise and interference compared to a balanced connection.

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  • 5. 

    A "TT" or Tiny Telephone jack comes in TS and TRS forms?

    • A.

      True

    • B.

      False

    Correct Answer
    A. True
    Explanation
    A "TT" or Tiny Telephone jack does come in TS (Tip-Sleeve) and TRS (Tip-Ring-Sleeve) forms. The TS form has two conductors, one for the signal (tip) and one for the ground (sleeve), while the TRS form has an additional conductor for stereo signals or balanced audio. Both forms are commonly used in audio equipment and telecommunications devices.

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  • 6. 

    Which of these answers describes two forms of balanced connector cables? Please select the best answer to the question:

    • A.

      TRS, TT-TS,

    • B.

      TS & TT-TRS

    • C.

      TRS, TT-TRS

    • D.

      TS & TT-TS

    Correct Answer
    C. TRS, TT-TRS
    Explanation
    The answer TRS, TT-TRS describes two forms of balanced connector cables. TRS stands for Tip-Ring-Sleeve, which is a type of connector that has three conductors for carrying the audio signal. TT-TRS refers to a connector that has two conductors for carrying the audio signal. Both TRS and TT-TRS connectors are commonly used in audio applications to ensure a balanced signal transmission, which helps to reduce noise and interference.

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  • 7. 

    A small inexpensive coaxial connector used for interconnection of many audio devices, especially consumer devices. The coaxial configuration (a center "hot" conductor surrounded by a "ground" connection that is a consistent distance from the hot) also makes them an affordable solution for high frequency transmission of signals like video and digital audio so long as proper insulators are used that will maintain the proper impedance through the connector. S/PDIF formatted digital audio data is often transmitted on coaxial cable terminated with this type of connector. This connection is called a _____________ cable.

    Correct Answer
    phono plug
    phono
    RCA
    rca
    Explanation
    The connection described in the question is commonly known as a phono plug, phono, RCA, or rca cable. It is a small coaxial connector used for interconnecting audio devices, particularly consumer devices. The coaxial configuration of the cable allows for affordable transmission of high-frequency signals like video and digital audio, as long as proper insulators are used to maintain the correct impedance. S/PDIF formatted digital audio data is often transmitted using this type of connector.

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  • 8. 

    AES/EBU is the most common alternative to the S/PDIF standard and the most common AES/EBU physical interconnect is AES Type I Balanced - 3 conductor, 110 ohm twisted pair cabling with an XLR connector.

    • A.

      True

    • B.

      False

    Correct Answer
    A. True
    Explanation
    The statement is true because AES/EBU is indeed the most common alternative to the S/PDIF standard. The AES/EBU physical interconnect that is commonly used is AES Type I Balanced, which consists of 3 conductor, 110 ohm twisted pair cabling with an XLR connector.

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  • 9. 

    S/PDIF typically uses either balanced, low impedance coaxial cables or fiber optic cables for transmission. When using coaxial cables for transmission, it is normally best to keep cable lengths to a minimum, and to use the best quality 75 ohm video-type cables available.

    • A.

      True

    • B.

      False

    Correct Answer
    B. False
    Explanation
    S/PDIF typically uses either unbalanced, high impedance coaxial cables or fiber optic cables for transmission. When using coaxial cables for transmission, it is normally best to keep cable lengths to a minimum, and to use the best quality 75 ohm video-type cables available.

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  • 10. 

    What is a BNC Connector and what is it used for?

    • A.

      A type of coaxial connector. In audio gear, BNC connectors are normally used to carry synchronizing clock signals between devices. BNCs are bayonet-type connectors, rather than screw on, or straight plugs. They are named for their type (Bayonet), and their inventor, Neill Concelman.

    • B.

      A type of coaxial connector. In audio gear, BNC connectors are normally used to transfer MIDI protocol between an instrument and the DAW station. BNCs are bayonet-type connectors, rather than screw on, or straight plugs. They are named for their type (Bayonet), and their inventor, Neill Concelman.

    • C.

      A type of XLR connection. In audio gear, BNC connectors are normally used to carry synchronizing clock signals between devices. BNCs are bayonet-type connectors, rather than screw on, or straight plugs. They are named for their type (Bayonet), and their inventor, Neill Concelman.

    Correct Answer
    A. A type of coaxial connector. In audio gear, BNC connectors are normally used to carry synchronizing clock signals between devices. BNCs are bayonet-type connectors, rather than screw on, or straight plugs. They are named for their type (Bayonet), and their inventor, Neill Concelman.
    Explanation
    BNC connectors are a type of coaxial connector commonly used in audio gear to carry synchronizing clock signals between devices. They are bayonet-type connectors, meaning they can be easily and quickly connected and disconnected without the need for screwing or straight plugs. The name "BNC" stands for "Bayonet Neill Concelman," after their inventor.

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  • 11. 

    This type of connector has gone through several generations and is currently a 20-bit digital format. This optical connection is used for transferring digital data 8-tracks at a time and has become a standard of the industry.

    • A.

      TT

    • B.

      XLR

    • C.

      S/PDIF

    • D.

      ADAT

    Correct Answer
    D. ADAT
    Explanation
    ADAT stands for Alesis Digital Audio Tape, which is a type of optical connector used for transferring digital audio data. It has gone through several generations and is currently a 20-bit digital format. ADAT allows for the transfer of 8 tracks of digital audio data at a time, making it a popular choice in the industry. It has become a standard for many recording studios and audio professionals.

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  • 12. 

    Is this connector blanced or unbalanced?

    • A.

      Unbalanced

    • B.

      Balanced

    Correct Answer
    B. Balanced
    Explanation
    x. Balanced Connection – In audio, the opposite of Unbalanced. For us balanced refers to a type of AC electrical signal having two "legs" independent of ground. One is generally considered positive (+) and the other negative (-) in voltage and current flow with respect to ground. Unlike unbalanced audio lines there is no "signal" carried in the shield or ground connection unless there is a fault. The main benefit is that any noise that gets induced into the line will be common to both the positive and negative sides and thus canceled when it arrives at its destination, assuming the destination is balanced. This phenomenon is called "Common Mode Rejection" and basically just means that any signals common to both the positive and negative legs of balanced lines get canceled. This happens because when the receiving device looks at the signal the common noise actually shows up as out of phase with itself, and gets cancelled. Think of it as if the negative (-) signal gets inverted to positive (+) before use, which puts the desired audio signal in phase with the already positive other leg and at the same time causes the undesired common noise to become out of phase with itself. Clear as mud? Balanced lines are generally much better for long cable runs due to their ability to reject induced noises. XLR and TRS type cables are designed to transmit balanced audio from one balanced device to another. A standard 1/4-inch guitar cable is an example of an unbalanced cable. Another (newer) application of balancing that is becoming popular in audio systems is the idea of balanced power systems. Fundamentally the concept is the same. There is a positive and negative (with respect to ground) leg of electricity at every electrical outlet. The idea is that the power supply of any devices connected can then reject any noise induced on the AC line and thus will produce cleaner audio. We'll talk more about balanced AC systems in the future.

    xi. Unbalanced Connection – In electronics, a condition where the two legs of the circuit are unbalanced with respect to ground, usually because one leg is kept at ground potential. In other words: An audio signal requires two wires or conductors to function. In an unbalanced situation, one of those conductors is used to carry both signal and ground (shield). Unbalanced circuits tend to be less expensive to construct, but they are much more susceptible to induced noise problems than their balanced counterparts. This is because any induced noise in one conductor is not canceled by similar noise in the other conductor (as in a balanced line) and may be carried with signal into connected equipment. In general, unbalanced lines should be kept as short as possible (certainly under 25-30' maximum) to minimize potential noise problems.

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  • 13. 

    Line Level – Literally, the average voltage of an electronic audio signal. While technically any voltage over 25 millivolts RMS is considered line level, in the modern audio world we narrow the scope a bit to the two line level references in use today: Balanced "pro" gear runs at around ______ dBm (1.23 volts), while unbalanced "semi-pro" gear operates at approximately .316 volts (-10 dBV). "Pro" and "semi-pro" may be almost meaningless terms anymore, but the two operating levels must still be dealt with.

    Correct Answer
    +4
    4
    four
    Four
    Explanation
    The correct answer is "+4". In the modern audio world, balanced "pro" gear operates at around +4 dBm, which is equivalent to 1.23 volts. This is one of the line level references in use today.

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  • 14. 

    Mic Level – The level (or voltage) of signal generated by a microphone. Typically around 4 millivolts. Compare this with the two normal line levels (1.23 volts and .316 volts), and it becomes apparent just how much amplification is going on in a microphone preamp, and why it is essential that preamps be of as high quality as possible!

    • A.

      True

    • B.

      False

    Correct Answer
    B. False
    Explanation
    Mic Level is around 2 millivolts, not 4.

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  • 15. 

    There is no standard for instrument level; it is assumed to fall between mic level (on the low end) and line level (on the high end), and it can range from a few millivolts for passive or piezo pickups to several volts on instruments with active pickups and built-in preamps.

    • A.

      True

    • B.

      False

    Correct Answer
    A. True
    Explanation
    The statement is true because there is indeed no standard for instrument level. It is generally understood to fall between mic level and line level, and the actual level can vary depending on the type of instrument and its pickups. Passive or piezo pickups may have a lower level in the millivolt range, while instruments with active pickups and built-in preamps can have higher levels in the volt range.

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  • 16. 

    The__________ microphone is a very simple mechanical system, with almost no moving parts compared to other microphone designs. It is also one of the oldest microphone types, dating back to the early 1900's. It is simply a thin stretched conductive diaphragm held close to a metal disk called a backplate. This arrangement basically produces a capacitor, and is given its electric charge by an external voltage source. When sound pressure acts on the diaphragm it vibrates slightly in response to the waveform. This causes the capacitance to vary in a like manner, which causes a variance in its output voltage. This voltage variation is the signal output of the microphone.

    Correct Answer
    condenser
    Condenser
    Explanation
    A condenser microphone is a type of microphone that has a very simple mechanical system with few moving parts. It consists of a thin stretched conductive diaphragm held close to a metal disk called a backplate, creating a capacitor. An external voltage source charges the capacitor, and when sound pressure acts on the diaphragm, it vibrates, causing the capacitance to vary. This variation in capacitance results in a voltage variation, which is the signal output of the microphone. Condenser microphones have been in use since the early 1900s and are one of the oldest microphone types.

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  • 17. 

    A ________________ mic is one in which audio signal is generated by the motion of a conductor within a magnetic field. ,A very thin, light, diaphragm moves in response to sound pressure. The diaphragm's motion causes a voice coil which is suspended in a magnetic field to move, generating a small electric current. This type of microphone is robust in construction and can often handle very high SPLs (Sound Pressure Levels). This mic does not require an external power source to operate. Because of the mechanical nature of their operation these mics are commonly less sensitive to transients, and may not reproduce quite the high frequency "detail" other types of mics can produce. This type of mic is commonly used in live applications. In the studio, they are often used to record electric guitar and drums.

    Correct Answer
    dynamic
    Dynamic
    Explanation
    A dynamic microphone is one in which audio signal is generated by the motion of a conductor within a magnetic field. It consists of a thin, light diaphragm that moves in response to sound pressure. This motion causes a voice coil, suspended in a magnetic field, to move and generate a small electric current. Dynamic microphones are robust in construction and can handle high sound pressure levels without the need for an external power source. However, due to their mechanical operation, they may be less sensitive to transients and may not reproduce high-frequency detail as accurately as other types of microphones. They are commonly used in live applications and for recording electric guitar and drums in the studio.

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  • 18. 

    Many of our recent Piano Miking Suggestions recommended use of _________ microphones. These microphones use a small electret capsule mounted close to a backing plate. The idea is that the mic capsule/plate is mounted to a large flat surface (or boundary). This increases the sensitivity of the mic by 6 dB (due to pressure doubling from reflected soundwaves), and gives it a hemispherical pickup pattern. The practical frequency response of the mic will depend on the size of the flat surface it is mounted to. If the surface is too small, low frequencies will not be reflected resulting in an apparent high frequency (treble) boost.

    Correct Answer
    pzm
    PZM
    boundary
    Boundary
    Explanation
    The correct answer is "pzm, PZM, boundary, Boundary". In recent Piano Miking Suggestions, the recommendation is to use pzm or boundary microphones. These microphones have a small electret capsule mounted close to a backing plate and are mounted on a large flat surface. This setup increases the sensitivity of the microphone by 6 dB due to the doubling of pressure from reflected soundwaves and gives it a hemispherical pickup pattern. The size of the flat surface affects the practical frequency response of the microphone, with a smaller surface resulting in a high frequency boost.

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  • 19. 

    A type of velocity microphone. A velocity microphone responds to the velocity of air molecules passing it rather than the Sound Pressure Level, which is what most other microphones respond to. In many cases this functional difference isn't important, but it can certainly be an issue on a windy day. Very old __________ mics could be destroyed from the air velocity created just by carrying them across a room. A ribbon mic works by loosely suspending a small element (usually a corrugated strip of metal) in a strong magnetic field. This strip is moved by the action of air molecules and when it moves it cuts across the magnetic lines of flux causing a signal to be generated. Naturally this type of mic has a figure 8 pick up pattern. You can think of it like a window blind; it is easily moved by wind blowing at it, but usually doesn't move when wind blows across it from left to right.

    Correct Answer
    ribbon
    Ribbon
    Explanation
    The correct answer is "ribbon" or "Ribbon". This is because the passage describes a type of microphone that is easily affected by air velocity, specifically on a windy day. The microphone described is a ribbon microphone, which works by suspending a small strip of metal in a magnetic field. When air molecules move the strip, it generates a signal. The analogy of a window blind is used to explain that the ribbon microphone is easily moved by wind blowing at it, but not when wind blows across it from left to right.

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  • 20. 

    .The difference in sound quality due to different microphone preamp models has to do with different inpedences produced by different microphones.

    • A.

      True

    • B.

      False

    Correct Answer
    A. True
    Explanation
    Impedance - Measured in ohms, impedance refers to the resistance of a circuit or device to AC (alternating current). Such an AC circuit could be any two audio devices connected together, like a speaker and an amp, passing audio signals. All other things being equal, more power (watts) will flow through a speaker with a low impedance than one with a high impedance. This will also put a greater strain on the amplifier to try to produce this power. If the impedance is too low your amp will not be able to handle it and bad things will happen. Most modern electronic audio devices have extremely high input impedances so they can be driven by very low power outputs. This is one of many reasons why high quality audio equipment can be built so much less expensively these days

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  • 21. 

    A converter used to transfer signal from a microphone to a digital DAW processes the signal with an __________ to __________ converter. Please type your answer in lower case with a comma and space between the words: Example:  radio, television

    Correct Answer
    analog, digital
    Analog, Digital
    Explanation
    The converter used to transfer signal from a microphone to a digital DAW processes the signal with an analog to digital converter.

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  • 22. 

    In a lossy compression scheme, such as MPEG video used in DVD production, the quality of the audio and video may be degraded somewhat. Another distinction is whether or not a special decoding algorithm is required to access or uncompress the file. Usually (but certainly not always) lossless compression schemes require special decoding software to restore the file to its original form, whereas lossy compression schemes are often encoded so they can be accessed without the extra decompression step.

    • A.

      True

    • B.

      False

    Correct Answer
    A. True
    Explanation
    Lossy compression schemes, like MPEG video used in DVD production, do indeed degrade the quality of audio and video to some extent. Additionally, lossless compression schemes typically require special decoding software to restore the file to its original form, while lossy compression schemes are often encoded in a way that allows them to be accessed without the need for extra decompression. Therefore, the statement "In a lossy compression scheme, such as MPEG video used in DVD production, the quality of the audio and video may be degraded somewhat" is true.

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  • 23. 

    What is the primary difference between a compressor and a limiter?

    • A.

      The primary difference is the ratio used in reducing gain. In a limiter, this ratio is set up to be as close to infinity:1 as possible (no matter how much the input signal changes, the output level should remain pretty much constant)

    • B.

      The primary difference is the ratio used in increase gain. In a limiter, this ratio is set up to be as close to infinity:1 as possible (no matter how much the input signal changes, the output level should increase exponentially)

    • C.

      The primary difference is the ratio used in reducing gain. In a limiter, this ratio is set up to be as limited as possible (no matter how much the input signal changes, the output level should remain pretty much constant)

    Correct Answer
    A. The primary difference is the ratio used in reducing gain. In a limiter, this ratio is set up to be as close to infinity:1 as possible (no matter how much the input signal changes, the output level should remain pretty much constant)
    Explanation
    A limiter and a compressor both reduce gain, but the primary difference lies in the ratio used to reduce the gain. In a limiter, the ratio is set to be as close to infinity:1 as possible. This means that no matter how much the input signal changes, the output level will remain relatively constant. In contrast, a compressor does not have such a high ratio and allows for more dynamic variation in the output level.

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  • 24. 

    What is a De-Esser and what is it used for?

    • A.

      A dynamics processor that establishes a maximum gain setting and prevents signals from getting any louder than that setting.

    • B.

      – A dynamics device whose function is to remove unwanted audio material below a certain threshold. Some type of "gain cell" is employed (usually a VCA) that can raise or lower the volume of the audio going through the unit. When the signal falls below a certain threshold that is set the gain cell will quickly drop the audio level down to a predetermined level.

    • C.

      A special type of compressor that is tuned to be sensitive to sibilant sounds, or sounds with high frequencies such as the sound produced by the letter "s." The need for this process arises out of a combination of the presence peak many microphones have in their frequency response to accentuate vocal recording combined with close proximity vocal work and possible added high frequency boost from equalizers and tone controls.

    Correct Answer
    C. A special type of compressor that is tuned to be sensitive to sibilant sounds, or sounds with high frequencies such as the sound produced by the letter "s." The need for this process arises out of a combination of the presence peak many microphones have in their frequency response to accentuate vocal recording combined with close proximity vocal work and possible added high frequency boost from equalizers and tone controls.
    Explanation
    DeEsser – A special type of compressor that is tuned to be sensitive to sibilant sounds, or sounds with high frequencies such as the sound produced by the letter "s", hence the name de-esser. The need for de-essing arises out of a combination of the presence peak many microphones have in their frequency response to accentuate vocal recording combined with close proximity vocal work and possible added high frequency boost from equalizers and tone controls. While these things often make a vocal track have more "air" and high-end clarity, they can also add enough accentuation to certain consonants (especially the "s") that they become too pronounced. The problem can range from being slightly annoying to being bad enough to cause distortion in the signal path. Many years ago broadcast engineers figured out they could tune compressors to be more sensitive to these frequencies, which in effect produces an automatic volume control that can turn down the audio anytime one of the sibilant sounds occur. In fact, any compressor with a sidechain input can be turned into a de-esser by inserting an EQ and boosting the offending frequencies. Even more flexibility comes from using a multi-band compressor. The de-essing action no longer has to lower the overall signal level. It can just lower the level in the specific range of frequencies specified. Some modern de-essers, however, have very sophisticated circuitry and controls that are optimized for achieving results beyond what would be easy with a simple compressor with an EQ in the sidechain.

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  • 25. 

    The remainder of sound that exists in a room after the source of the sound has stopped is called ______________

    • A.

      Echo

    • B.

      Reverb

    • C.

      Delay

    Correct Answer
    B. Reverb
    Explanation
    Reverb refers to the remainder of sound that exists in a room after the source of the sound has stopped. It is caused by sound waves reflecting off surfaces such as walls, floors, and ceilings. This reflection creates a series of multiple echoes which blend together to create a complex, lingering sound. Reverb is commonly used in music production and sound engineering to add depth and richness to audio recordings.

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  • 26. 

    The processof creating delays in audio production tape machines to create a comb filter effect is called flanging.

    • A.

      True

    • B.

      False

    Correct Answer
    A. True
    Explanation
    Flange – A flange is the metal rim or the reel part of a reel to reel tape (also called open reel tape), as opposed to the hub. Years ago when tape machines were used to create delays in audio production a process called flanging was invented. It consisted of recording the same signal on two tapes each playing together and then, using pressure to one of the reel flanges, briefly slowing down one of the machines. The short timing discrepancies that result produce a very pronounced comb filter effect. The effect was often modulated by alternating pressure to each machine's reels. One machine would slow down relative to the other, and then the second machine would be slowed beyond the first. It was also possible to route some of the signal being played back into the recording circuit to provide regeneration and resonance effects. Later electronic flangers were invented that used a modulated analog or digital delay line, which was mixed back with the dry signal. While much more convenient than the old open reel approach many engineers agree that the electronic units have never sounded as good as the "reel" thing.

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  • 27. 

    In regards to on-axis microphone technique, which picture below describes correct placement: (A) (B)

    • A.

      A

    • B.

      B

    Correct Answer
    A. A
    Explanation
    Picture A shows the correct placement for on-axis microphone technique. The microphone is positioned directly in front of the sound source, ensuring that it captures the sound directly and accurately. This placement helps to minimize any off-axis sound and unwanted noise, resulting in a clearer recording or sound reinforcement. Picture B, on the other hand, shows an incorrect placement as the microphone is positioned at an angle to the sound source, which can result in a less accurate and distorted sound capture.

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  • 28. 

    A ________ filter is a filter that has a series of very deep notches in its frequency response with the spacing of all of the notches at multiples of the frequency of the lowest notch (they are all harmonically related). It got its name from looking like a comb when plotted on a frequency response graph. These filters are produced when a signal is time delayed and added back to itself. Some frequencies will cancel and others will be reinforced, which can dramatically change the tonal color of the sound. In practice this is common problem that occurs when a stereo mix is collapsed to mono because many stereo effects, such as chorus and flanging, achieve their stereo imaging by using some form of the Haas effect.

    Correct Answer
    comb
    Comb
    Explanation
    A comb filter is a type of filter that has multiple notches in its frequency response, with the spacing between the notches being multiples of the frequency of the lowest notch. When plotted on a frequency response graph, it resembles the shape of a comb. These filters are created by time delaying and adding a signal back to itself, causing certain frequencies to cancel out and others to be reinforced. This can have a significant impact on the tonal color of the sound. The term "comb" or "Comb" is used to describe this type of filter.

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  • 29. 

    In regards to phasing, two identical waveforms, 180 degrees out of phase, will produce a "hollow" sound, but not cancel out completely.

    • A.

      True

    • B.

      False

    Correct Answer
    B. False
    Explanation
    Phasing – Audio waveforms are cyclical; that is, they proceed through regular cycles or repetitions. Phase is defined as how far along its cycle a given waveform is. The measurement of phase is given in degrees, with 360 degrees being one complete cycle. One concern with phase becomes apparent when mixing together two waveforms. If these waveform are "out of phase", or delayed with respect to one another, there will be some cancellation in the resulting audio. This often produces what is described as a "hollow" sound. How much cancellation, and which frequencies it occurs at depends on the waveforms involved, and how far out of phase they are (two identical waveforms, 180 degrees out of phase, will cancel completely).

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  • 30. 

    What is flutter echo and how is it caused?

    • A.

      A condition that occurs outdoors when two parallel waveforms cancel each other out and leave a distinct reverberation to listeners

    • B.

      A condition that occurs in acoustic spaces when two parallel surfaces reflecting sound between one another are far enough apart that a listener hears the reflections between them as distinct echoes.

    • C.

      An effect in smaller rooms that is caused by phasing. The resulting sound is a loud hum within solid surfaces.

    Correct Answer
    B. A condition that occurs in acoustic spaces when two parallel surfaces reflecting sound between one another are far enough apart that a listener hears the reflections between them as distinct echoes.
    Explanation
    Flutter Echo – A condition that occurs in acoustic spaces when two parallel surfaces reflecting sound between one another are far enough apart that a listener hears the reflections between them as distinct echoes. The audible effect is in many cases a sort of "fluttering" sound as the echoes occur in rapid succession. In smaller rooms it can take on a sort of tube-like hollow sound, as the echoes are closer together.

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  • 31. 

    Define Lossless Data Compression:

    • A.

      A data compression procedure that reduces the size of (encodes) digital audio files without sacrificing any audio data, or fidelity, when the files are expanded (decoded) for playback

    • B.

      A data compression procedure which uses sophisticated algorithms to discard selected bits contained in the original audio that have a minimal impact on the overall sound

    Correct Answer
    A. A data compression procedure that reduces the size of (encodes) digital audio files without sacrificing any audio data, or fidelity, when the files are expanded (decoded) for playback
    Explanation
    Lossless data compression refers to a procedure that reduces the size of digital audio files without compromising any audio data or fidelity. This means that when the compressed files are expanded or decoded for playback, they retain the exact same quality as the original uncompressed files. This compression technique does not discard any audio data or bits, ensuring that the sound remains intact.

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  • 32. 

    _____________ is noise added intentionally to a digital recording. Low level signals are difficult for digital gear to record; the sampling machine simply has difficulty deciding whether the necessary bits should be turned on or off, creating "quantization noise." By adding a small amount of very controlled noise to the original signal, the bits can be made to positively switch on or off, improving low level sound resolution. The noise used is often "shaped" to be in-offensive to human ears.

    Correct Answer
    dither
    Dither
    Explanation
    Dither is noise intentionally added to a digital recording to improve the resolution of low level signals. When recording low level signals, the sampling machine may have difficulty deciding whether the necessary bits should be turned on or off, resulting in quantization noise. By adding a small amount of controlled noise to the original signal, the bits can be made to positively switch on or off, enhancing the clarity of low level sound. This added noise, known as dither, is often shaped to be inoffensive to human ears.

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Our quizzes are rigorously reviewed, monitored and continuously updated by our expert board to maintain accuracy, relevance, and timeliness.

  • Current Version
  • Sep 07, 2023
    Quiz Edited by
    ProProfs Editorial Team
  • May 11, 2011
    Quiz Created by
    Amcgrane
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