1.2 Proximity Effect

  • The proximity effect is when low frequencies are emphasised when you get closer to the microphone
  • This can be fixed by moving further back or by using EQ later on. Sometimes this effect is used to give a ‘warmer’ sound
  • It occurs in directional (cardioid, hypercardioid or figure of 8) microphones
  • The diagram below shows the proximity effect at different distances away from the microphone.

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1.1 Hardware: DI Boxes

DI Boxes

  •  DI boxes (direct input) are used to eliminate the need to mic up electronic instruments.
  • They can be plugged directly in to a mixer or audio interface.
  • They convert an unbalanced (two wire) signal to a balanced (three wire)
  • Microphones usually use XLR connections (Extra Low Resistance).
  • Balanced cables use phase cancellation to reduce noise; they copy and invert the sound signal at both ends then use the resulting signal to cancel out noise.

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Welcome to StudyMusicTech!

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Welcome to the StudyMusicTech website! This site is designed to help you study for the Pearson Edexcel A Level Music Technology qualification for first teaching in 2017. Whether you are a teacher or student, this website is to help you prepare for the examinations, and is written by a head of department, examiner and trainer.

Use the links on the left to get around, and click on ‘Tutorials’ to view the content.

1.1 Hardware: Audio Interfaces

The picture below shows a simple FocusRite audio interface with two separate inputs.

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  • Latency can be an issue when using external audio interfaces. This is when there is a delay between playing the signal and either the computer recording it or hearing it being monitored back.
  • This is very off putting when performing and can mean that the ‘feel’ or ‘groove’ of the song is lost and even the performance ending up out of time
  • Audio quantise functions can sometimes help to improve this, as can reducing the buffer size on your DAW
  • CD quality recordings use a sample rate of 44.1kHz and a bit depth of 16 bits.
  • Even some relatively inexpensive audio interfaces can record at higher bitrates, such as 48kHz and 24 bit.
  • The Red Book Standard outlines the requirements for digital audio and CDs
  • A higher sample rate improves the capture of higher frequencies and thus the higher frequency response. Nyquist’s theorem states that the highest frequency captured is half of the sample rate, and thus to capture the human hearing range, a sample rate of at least 40kHz must be used.
  • Benefits of using a higher bit depth include capture of wider dynamic ranges and minimising noise.
  • In order to capture or play back digital audio, some conversion needs to take place. Thus, audio interfaces incorporate analogue to digital converters (ADCs) and digital to analogue converters (DACs)
  • Audio interfaces will normally incorporate some kind of meter. This shows the volume of the input or output signal and allows the operator to avoid distortion.

 

XLR / Jack / Combi Inputs

·      There are two combi inputs. These can take both XLR connections and jack connections.

·      Jack connections can be TS (tip/sleeve) and TRS (tip/ring/sleeve).

·      XLR connections are balanced. This reduces noise using phase cancellation from combining two signals with opposite polarity and the resulting destructive interference

·      XLR connections are often used for microphones; jack connections for electric guitars and synthesisers

·      There is a locking tab at the top of the connection which means that the XLR cable does not come loose when pulled without pressing the locking tab

·      A pre-amp will amplify the input signal as part of the audio interface

Gain / Level / Pad

·      For each input, you can select between instrument and line level.

·      The gain control is used to set the input for a good signal to noise ratio and thus reduce noise and prevent distortion

·      The pad switch reduces the sensitivity in order to prevent distortion if the signal being recorded is loud. Pad switches often reduce the input signal by up to 20dB.

Phantom Power / LED Indicators

·      The 48V switch activates or deactivates phantom power.

·      This is used to supply power to a condenser microphone or active DI box

·      The interface only has one switch so this controls phantom power for all channels

·      This would create issues if using a condenser and a ribbon microphone as phantom power would break a ribbon microphone

·      MIDI stands for Musical Instrument Digital Interface

·      It is used as a language to communicate between different electronic instruments

·      The LED identifies when a MIDI signal is being sent or received

·      The second LED identifies when a USB connection to a computer is active

Headphones

·      Headphones can be used to monitor the signal

·      The headphone socket is a TRS/stereo jack connection. It can be used to provide a monitor mix (this may be different to what would be coming out of the monitor speakers).

·      On this interface, you can change which output the monitor output mirrors, and use the knob to change the volume of the headphones to ensure a good monitor level and avoid damaging your hearing

USB Connector

·      USB stands for Universal Serial Bus

·      It allows the interface to be connected to a computer and is used for transferring data

·      Power can be provided on a USB cable

·      USB 2.0 is a development of the original USB protocol, but is not as fast as ThunderBolt (Apple) and FireWire (IEEE 1394) connections.

MIDI Ports

·      These connectors are 5 pin DIN-180 connectors

·      They can be used to connect other equipment such as synthesisers and effects units

·      Equipment is connected together in a loop, with the MIDI out of one piece of equipment being connected to the MIDI in of the next piece

·      Some interfaces incorporate a MIDI-thru connector, which reduces the latency on the MIDI network

Outputs

·      All sets of outputs here are in stereo, and are analogue

·      There are two sets of unbalanced phono / RCA connections.

·      The red connector is for the right channel and the white for the left

·      Unbalanced connectors are are more susceptible to interference and have a poorer signal-to-noise ratio than balanced connectors

·      The two jack outputs are balanced; they are using a TRS jack connector

 

3.3 Leslie Speaker / Rotary Effect

Rotary Speaker

  • The Leslie speaker is a combined amplifier and loudspeaker that modifies the sound of an instrument as well as amplifying it, by rotating sound waves. It is most commonly associated with the Hammond organ, though it was later used for the guitar and other instruments.
  • A typical Leslie speaker contains an amplifier, treble and bass speaker, though specific components depend on the model. Control is achieved either by an external half-moon switch, or by a foot pedal, that alternates between two settings known as “chorale” and “tremolo”.
  • The speaker is named after its inventor, Donald Leslie. Leslie began working in the late 1930s to get a speaker for a Hammond organ that had a closer emulation of a pipe or theatre organ, and discovered that rotating sound gave the best effect. Hammond was not interested in marketing or selling the speakers, so Leslie sold them himself as an add-on, targeting other organs as well as Hammond. The first speaker was made in 1941. The sound of the organ being played through his speakers received national radio exposure across the US, and it became a commercial and critical success as an essential part of any jazz organist.
  • Because the Leslie is a sound modification device in its own right, various attempts have been made to emulate the effect using electronics. The Univox Uni-Vibe was used by a number of notable musicians, while the Neo Ventilator has received critical praise, and Hammond-Suzuki now manufacture their own simulator in a box.
  • A Leslie speaker consists of a number of individual components. The audio signal enters the amplifier from the instrument. Once amplified, the signal travels to an audio crossover, where it is split into separate frequency bands that can be individually routed to each loudspeaker.
  • Different models have different combinations of speakers, but the most common model, the 122, consists of a single woofer for bass and tweeter for treble. The audio emitted by the speakers is isolated inside an enclosure, aside from a number of outlets which lead towards either a rotating horn or drum. Both the horn and the drum are rotated at a constant speed by an electric motor.
  • The only control common to all Leslie speakers is a dial controlling the master volume. This is normally set up once and then left, since the volume is designed to be controlled by the organ’s expression pedal. Leslie recommended playing the organ at full volume with all stops or drawbars pulled out and adjusting the volume just before distortion occurs. However, the distorted sound of an overdriven vacuum tube amplifier can be a desirable sound, to the extent that modern Leslie simulators have an explicit “overdrive” setting. The half-moon switch on a Hammond organ that changes setting on the Leslie speaker between “chorale” and “tremolo”
  • Unlike most popular music amplifiers, that use jack plugs to connect to instruments, Leslie speakers use an amphenol connector to interface directly to an organ via a console connector. Older models that used tube power amplifiers used a variety of 6-pin connectors, while later models used a 9-pin connector. In all cases, for a single organ – Leslie configuration, the mains power, audio and control signals are all carried on the connector, and the design of the pin layouts varies between organs and speakers. It is also possible to connect multiple Leslie speakers to a single organ, by using a power relay that provides the necessary AC current. A separate device known as the combo preamp is necessary to connect a vintage Leslie to another instrument such as a guitar. This combines a separate AC input and line level input onto a single amphenol connector, and provide a footswitch to select between the speeds of the Leslie.
  • The Leslie is specifically designed, via reproduction of the Doppler effect, to alter or modify sound. As the sound source is rotated around a specific pivot point, it produces tremolo (the modulation of amplitude) and a variation in pitch. This produces a sequence of frequency modulated sidebands. To stop a Leslie’s rotor, a special brake circuit was added to the Leslie motor controls, that incorporated an electronic relay by producing a half-wave of direct current.
  • Much of the Leslie’s unique tone is due to the fact that the system is at least partially enclosed, whereby linear louvres along the sides and front of the unit can vent the sound from within the box after the sound has bounced around inside, mellowing it. The crossover is deliberately set to 800 Hz to give the optimum balance between the horn and the drum, and is considered an integral part of the speaker. The tone is also affected by the wood used. Tone differences, due to cost cutting using particle board for speaker and rotor shelves instead of the previous plywood, are evident in the Leslie’s sound. The thinner ply of the top of the cabinet adds a certain resonance as well. Like an acoustic instrument, a Leslie’s tone is uniquely defined by its cabinet design and construction, the amplifier, crossover and speakers used, and the motors — not merely by the spinning of rotors.
  • Because a Leslie speaker modifies as well as amplifies the sound, the output cannot simply be connected to a larger PA system if the volume onstage from the built-in amplifier is too quiet. This is particularly problematic for an older Leslie like the 122 or 147, which only has a 40 watt RMS power amplifier. Instead, microphones are placed around the Leslie, and the output from these is connected to the PA.
  • The Beatles first recorded using a Leslie during the sessions for Revolver in 1966. After John Lennon had asked for his voice to sound “as though I’m the Dalai Lama singing from the highest mountain top”, Abbey Road engineer Geoff Emerick rewired the input of the studio’s Leslie so a vocal microphone could be attached to it. Emerick used this setup to record Lennon’s vocal on the track “Tomorrow Never Knows” and claims the Beatles subsequently wanted to record everything through a Leslie. George Harrison played his guitar through a Leslie on “Lucy in the Sky with Diamonds” and “You Never Give Me Your Money”. The Beatles subsequently inspired other guitarists to use the speaker. Eric Clapton played through one on Cream’s song “Badge”, and David Gilmour used a similar setup when recording with Pink Floyd.

3.3 Tape Delay

Tape Delay

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  • The peak level will illuminate if the signal clips or distorts and the VU meter shows the input level in volume units.
  • The mic/instrument volume is the gain. This can be decreased to avoid clipping. It can be turned up to obtain a better signal to noise ratio.
  • The instrument volume is for a low impedance instrument such as guitar.
  • The mode selector is used to select different taps / patterns / rhythms / types of delay / number or volume of repeats / combinations of playback heads.
  • The bass and treble controls are used to modify the tone of the delay (not the dry signal). It is a shelving filter that adjusts the levels of low and high frequencies.
  • The reverb/echo volume is the wet/dry effect mix. “Straight” gives no echo at all. This is the gain, and has a spring reverb unit.
  • Repeat rate is the delay time / the amount of time between each repeat.
  • Intensity is the feedback amount/number of repeats. This is the gain.
  • The power switch should be used so that the unit should be switched off when not in use to preserve the life of the tape.
  • Echo normal or footswitch works as a bypass (to switch off the machine).
  • HML gives different output levels/volumes so that the unit can match the different signal levels required by different studio equipment.

3.3 Introducing Tapes

What a Tape Is Made Of

  • They contain two reels of magnetic tape. The tape is coated with iron oxide.
  • The sound quality of cassette tape is not as good as studio tape, with a reduced high frequency response because the tape moves slowly at 1 and 7/8 inches per second, and the tape is narrow.
  • The speed of the tape is controlled by the capstan or pinch roller.
  • Mono tapes have one track on each side (so two in total)
  • Stereo tapes have two tracks on each side (so four in total)
  • Multitrack tapes have four tracks but only play one way
  • Tape lengths are 60, 90 and 120 minutes, but 120 minute tapes often snapped
  • Tapes had erase protection tabs which meant that when broken, a tape could not be recorded on.
  • Tapes could be ‘dubbed’, which raised concerns about piracy, especially when a technology called ‘high speed dubbing’ was developed.

 

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Drawbacks of Tape

  • Cassette tapes are prone to hiss, so Dolby Noise Reduction was developed. This boosts the high frequencies when recording and reduces them on playback.
  • Tape saturation occurs when an increase in the strength of the sound source and thus electrical signal cannot produce an equivalent increase in magnetisation on the tape head. This leads to a form of subtle compression.
  • Cassette tapes can stretch
  • Iron oxide wears off tape over time which means that an oxide build up can occur on the tape head
  • Thus many companies marketed tape head cleaning cassettes which, combined with a chemical, removed oxide from the tape heads.
  • Cassette tapes are prone to print through, which is where the music is heard as an echo before it plays
  • Tape has a leader tape, which cannot be recorded on. This lasts for 2-3 seconds and draws the tape through onto the other reel.