Write a couple good paragraphs on the SSL Duality console. Research it at the SSL website here - read the documentation, watch the videos, etc. Look also in Mix magazine or other online sources - do a Google search for "SSL Duality" and you should find plenty. Compare what the Duality has to what is discussed in Chapter 5 reading about consoles. Is it an analog or digital console? Does it have VU meters? Is there anything else about the console that interests you? etc.
Turn this in next week during class.
Monday, October 27, 2008
Sunday, October 26, 2008
Audio Consoles
Consoles (also known as boards or mixers) take input signals and amplify, balance, process, combine and route them for broadcasting or recording. Some consoles also store operational data.
There are a few broad categories of consoles: analog or digital, on-air broadcast or production/post-production, and software based virtual consoles.
Analog consoles use physical electric circuits and wiring to control that audio signals flow throughout the board.
The patch bay is used for flexibility in the signal flow. Jacks are wired to console components to allow for re-routing of signals both within the console as well as with other equipment outside of the board, such as an external Equalizer or Compressor unit. Patch cords are the cables used to plug into the jacks to direct signal flow.
Digital consoles convert the analog signal to digital at the input. Routing is done on board as well as through patching cords.
Virtual consoles use a simulated console on a computer screen with specialized software.
On-air broadcast consoles handle audio sources that are immediately available to an audience, along with pre-packaged audio for playback. It has more control for rapid transitions between the DJ, music, advertisements, guests, call-ins, and others.
Inputs bring the signals from the microphone, CD player, recorder, phone calls, etc. into the console. Faders control the loudness of each channel.
The master fader handles the combined signals feeding to it from the input channels and then outputs the combined final signal.
Features found on most consoles include the input/output channel strips, input selector (high versus low level signal), phantom power (48 V), mic pre amp (boosts weak mic signal to a voltage of useable level to minimize noise introduced by console), trim/gain (changes input sensitivites- boosts low level or prevents overload in high levels), pad (lowers power of signal when trim isn't enough), channel assignment and routing, pan pot (panoramic potentiometer - shifts proportion of sound to any point between L and R), EQ and filtering (alters signal's frequency response by raising or lowering level of a range of freencies or attenuating freq at certain levels), signal procession, FX sends to external signal processing, and Solo/Mute buttons.
There are also VU (Volume Unit) meters that indicate the loudness level based on a perceived volume, reading the average of the combined waveforms. Includes the percent of modulation in relation to the maximum of the system's ability as well as the dB level. Peak Meters are electronic instead using LEDs but have the same purpose as the VU.
Automation is used to store mixer settings for recall, in order to compare mix settings of a recording to one another. Write - creates automated mix where data is monitored and stored. Read - plays back automation data. Update - makes changes to automation.
Balanced cables are used because they resist electrical interference and therefore result in lower noise.
Digital consoles do not have an physical connections or circuits, all functions go through the central processor. This removes duplication of many channels, and increases the number of operations that can be done in a smaller area.
Digital Control Surfaces are similar to consoles but there are no audio signals in them, only control circuits that send digital instructions to other units for signal processing. These are used with Digital Audio Workstations (DAWs) like Pro Tools.
Helpful Links:
Cable information (XLR, TRS, etc)
There are a few broad categories of consoles: analog or digital, on-air broadcast or production/post-production, and software based virtual consoles.
Analog consoles use physical electric circuits and wiring to control that audio signals flow throughout the board.
The patch bay is used for flexibility in the signal flow. Jacks are wired to console components to allow for re-routing of signals both within the console as well as with other equipment outside of the board, such as an external Equalizer or Compressor unit. Patch cords are the cables used to plug into the jacks to direct signal flow.
Digital consoles convert the analog signal to digital at the input. Routing is done on board as well as through patching cords.
Virtual consoles use a simulated console on a computer screen with specialized software.
On-air broadcast consoles handle audio sources that are immediately available to an audience, along with pre-packaged audio for playback. It has more control for rapid transitions between the DJ, music, advertisements, guests, call-ins, and others.
Inputs bring the signals from the microphone, CD player, recorder, phone calls, etc. into the console. Faders control the loudness of each channel.
The master fader handles the combined signals feeding to it from the input channels and then outputs the combined final signal.
Features found on most consoles include the input/output channel strips, input selector (high versus low level signal), phantom power (48 V), mic pre amp (boosts weak mic signal to a voltage of useable level to minimize noise introduced by console), trim/gain (changes input sensitivites- boosts low level or prevents overload in high levels), pad (lowers power of signal when trim isn't enough), channel assignment and routing, pan pot (panoramic potentiometer - shifts proportion of sound to any point between L and R), EQ and filtering (alters signal's frequency response by raising or lowering level of a range of freencies or attenuating freq at certain levels), signal procession, FX sends to external signal processing, and Solo/Mute buttons.
There are also VU (Volume Unit) meters that indicate the loudness level based on a perceived volume, reading the average of the combined waveforms. Includes the percent of modulation in relation to the maximum of the system's ability as well as the dB level. Peak Meters are electronic instead using LEDs but have the same purpose as the VU.
Automation is used to store mixer settings for recall, in order to compare mix settings of a recording to one another. Write - creates automated mix where data is monitored and stored. Read - plays back automation data. Update - makes changes to automation.
Balanced cables are used because they resist electrical interference and therefore result in lower noise.
Digital consoles do not have an physical connections or circuits, all functions go through the central processor. This removes duplication of many channels, and increases the number of operations that can be done in a smaller area.
Digital Control Surfaces are similar to consoles but there are no audio signals in them, only control circuits that send digital instructions to other units for signal processing. These are used with Digital Audio Workstations (DAWs) like Pro Tools.
Helpful Links:
Cable information (XLR, TRS, etc)
Thursday, October 23, 2008
Music Industry information
Music Industry journal - interesting information on the latest developments in the music industry.
Do you have any other good music industry links? Please let me know!
Do you have any other good music industry links? Please let me know!
Wednesday, October 22, 2008
Tuesday, October 21, 2008
Microphones
A microphone is a transducer, which converts acoustic energy into electric energy. The element used in a mic names it.
Moving Coil (Dynamic) mics are generally more rugged, with less self-noise, and higher SPL without distortion, as well as less expensive. However the transducing element has more mass and therefore has a slower response to transients (quick attacks).
Ribbon mics are good for voice and music recording, but are more expensive and delicate. The transient response is better than the Dynamic.
Capacitor mics have the best transient response, but are also most epensive. They are good for distance miking and have low noise.
Polar Response Pattern is the direction a mic will receive sound. It is NOT related to the transducer in the mic.
Omnidirectional - all around the mic
Bidirectional - front and back
Unidirectional - front only
Cardioid - heart shaped
For all patterns, higher frequencies result in a more directional pattern, while lower frequencies have a less directional pattern.
Mic Sound Responses depend on:
- Magnetic Induction mics have a fixed magnet and a movable diaphragm.
- Moving Coil, or Dynamic, mics have a small lightweight coil in the magnetic field that moves in response to the sound waves hitting the diaphragm and creates a voltage proportional to the sound pressure level.
- Ribbon mics have a metal ribbon instead of a coil. The ribbon vibrates from the SPL and creates a voltage. Older ribbon mics used a vertical piece of metal, while newer ones use a horizontal ribbon which allows for the capsule to be smaller and lighter.
- Active Ribbon use an amplifier system that requires Phantom Power - remotely powering along an audio cable. This allows for a higher output and sound sensitivity.
- Variable Capacitance mics transduce energy using voltage variation instead of magnetic
- Capacitor mics have 2 parallel plates separated by a space . The front plate is a thin metalized plastic and the fixed back plate form a capacitor that holds charge. As sound hits the diaphragm, the capacitance changes and is translated into a relative level.
- Tube mics use a tube circuit to sound more present and livelier than the standard mic.
- Lavaliere mics use an electret diaphragm hat permanently holds a charge.
Moving Coil (Dynamic) mics are generally more rugged, with less self-noise, and higher SPL without distortion, as well as less expensive. However the transducing element has more mass and therefore has a slower response to transients (quick attacks).
Ribbon mics are good for voice and music recording, but are more expensive and delicate. The transient response is better than the Dynamic.
Capacitor mics have the best transient response, but are also most epensive. They are good for distance miking and have low noise.
Polar Response Pattern is the direction a mic will receive sound. It is NOT related to the transducer in the mic.
Omnidirectional - all around the mic
Bidirectional - front and back
Unidirectional - front only
Cardioid - heart shaped
For all patterns, higher frequencies result in a more directional pattern, while lower frequencies have a less directional pattern.
Mic Sound Responses depend on:
- Frequency response - the range of frequencies that a mic will reproduce at an equal level with a margin of 3 dB. Essentially the "coloration" of the mic.
- Overload - how well the mic handles the distortion if sound levels are too high. Ribbon mics are very sensitive to this and possibly damaged if overloaded.
- Max SPL - the level where output starts to distort - harmonics are introduced by the audio system that are not in the original signal. 120 dB = good, 135 dB = very good, 150 dB = excellent
- Sensitvity - measure the volltage the mic produces at SPL that indicates loudness. A capacitor has the loudest response, awhile ribbon has the quietest and requires more gain.
- Self Noise - the SPL that creates the same voltage as inherant electrical noise. 40 dB = fair, 30 dB = good, 20 dB or less = excellent
- Signal to Noise Ratio (SNR) - the difference between signal and noise levels in dB. [SPL/self noise] The larger the SNR, the less noise in the signal. 64dB = fair, 74 dB= good, 84 dB = excellent
- Proximity effect - bass frequencies increase when close to microphone. Pressure is raised at lower frequences, so Bass Roll Off is used to attenuate the bass frequencies and negate the effect.
- Hum - stray Alternating Current at 50-60 Hz occurs especially in dynamic mics. Use balanced cables to reduce this. Humbucking cancels induced hum with a 50/60Hz component.
Acoustics
Acoustics is the science of the sound wave's behavior. It is based upon the interaction of the sound waves in spaces with surfaces, where the sound is reflected, absorbed, or diffracted. Psychoacoustics is the study of how humans hear and perceive sound.
Direct sound is the first wave that reaches a listener's ears, directly from the sound source.
Early Reflections are the sound waves that bounce off the walls or other objects in the room and reach the listener withing 30 ms after the Direct sound. These Early reflections combine with the Direct sound to create the entire Initial sound.
Reverberant sound is comprised of the rest of the reflections as the amplitude and time between them decreases.
Early reflections add loudness and fullness to the direct sound and help define the subjective size of the room. They also hold the most energy of the sound.
Reverb Time (RT) is defined as the time it takes a sound to decrease 60 dB.
A longer RT gives the perception of a larger and hard-surface room, as well as being farther from the sound source. An RT of 1 sec or more is considered "live", while an RT of 1/2 sec or less is considered "dry".
An Echo is a distinct repetition of the Direct sound, and therefore occurs 35ms or more after the Direct sound, so that it will not be combined with the direct sound like early reverberations are.
The Haas, or precedence effect, occurs when two identical sound waves of the same intensity come from two separate sources, but one is delayed by 30-40ms. The sound emitted by the closest location is heard first. This creates the impression that the sound comes only from that closer location. The Haas effect is what allows us to perceive sounds as coming from between two stereo speakers.
Acoustic aspects:
Warmth - achieved with lower frequencies and increased RT
Intimacy - early reflections around 15-20 ms give a sense of closeness to the source
Clarity - the ability to hear sonic details, such as lyrics and attack of notes, when there is more direct sound than reverberant
Auditory Source Width (ASW) - the perception of the sounds width versus the actual, which is affected by the room, the microphone used, and the panning
Reverb Times used for:
Speech = 0.3-0.5 sec
Pop/Rock = 0.5 - 0.6 sec
Orchestral = 1.5 - 2 sec
Studio design is based on 4 factors: Sound Isolation, Room Dimensions, Room Shape, and Acoustics. All four have the ultimate goal of lowering noise - which is essentially any unwanted sound.
Noise Criteria (NC) is a rating system used to identify background noise levels. Similar to the Equal Loudness Curves, there is a graph that shows the dB needed for every frequency level and different NC values.
Isolation - Keep noise out and the sound source from going out of the studio.
Dimensions - The dimensions of a room add coloration to the recording, as certain frequencies are reinforced through resonance. Resonance occurs when a material has the same natural frequency as the sound wave and results in vibrating together in phase so that the amplitudes increase. Remove resonance in a room by creating Height X Width X Length ratios that are not multiples of one another (ie 10 X 20 X 30).
Shape - Parallel walls reinforce sound waves, since the angle the sound wave hits the wall equals the angle of reflection and bounces right back to the opposite side. To break up these parallel reflections, design the room with non-right angle corners or rounded walls.
Acoustics - Different surfaces interact with sound waves differently - they may absorb, reflect, diffract, or diffuse the sound. Absorption soaks up the sound, which results in a lifeless, dry sound. Reflection results in a complete bouncing off of the sound from the material, giving reverberation. Diffraction bends the sound around the surface, while diffusion reduces the amplitude of a wave and spreads it in time.
Link to more information on sound diffusion
Link to more information on sound diffraction
The Sound Absorption Coefficient goes from 1.0 for complete absorption to 0.0 for complete reflection. Different materials are assigned a value in this range based on how they react to sound.
Listen to interesting binaural microphone effects here
Direct sound is the first wave that reaches a listener's ears, directly from the sound source.
Early Reflections are the sound waves that bounce off the walls or other objects in the room and reach the listener withing 30 ms after the Direct sound. These Early reflections combine with the Direct sound to create the entire Initial sound.
Reverberant sound is comprised of the rest of the reflections as the amplitude and time between them decreases.
Early reflections add loudness and fullness to the direct sound and help define the subjective size of the room. They also hold the most energy of the sound.
Reverb Time (RT) is defined as the time it takes a sound to decrease 60 dB.
A longer RT gives the perception of a larger and hard-surface room, as well as being farther from the sound source. An RT of 1 sec or more is considered "live", while an RT of 1/2 sec or less is considered "dry".
An Echo is a distinct repetition of the Direct sound, and therefore occurs 35ms or more after the Direct sound, so that it will not be combined with the direct sound like early reverberations are.
The Haas, or precedence effect, occurs when two identical sound waves of the same intensity come from two separate sources, but one is delayed by 30-40ms. The sound emitted by the closest location is heard first. This creates the impression that the sound comes only from that closer location. The Haas effect is what allows us to perceive sounds as coming from between two stereo speakers.
Acoustic aspects:
Warmth - achieved with lower frequencies and increased RT
Intimacy - early reflections around 15-20 ms give a sense of closeness to the source
Clarity - the ability to hear sonic details, such as lyrics and attack of notes, when there is more direct sound than reverberant
Auditory Source Width (ASW) - the perception of the sounds width versus the actual, which is affected by the room, the microphone used, and the panning
Reverb Times used for:
Speech = 0.3-0.5 sec
Pop/Rock = 0.5 - 0.6 sec
Orchestral = 1.5 - 2 sec
Studio design is based on 4 factors: Sound Isolation, Room Dimensions, Room Shape, and Acoustics. All four have the ultimate goal of lowering noise - which is essentially any unwanted sound.
Noise Criteria (NC) is a rating system used to identify background noise levels. Similar to the Equal Loudness Curves, there is a graph that shows the dB needed for every frequency level and different NC values.
Isolation - Keep noise out and the sound source from going out of the studio.
Dimensions - The dimensions of a room add coloration to the recording, as certain frequencies are reinforced through resonance. Resonance occurs when a material has the same natural frequency as the sound wave and results in vibrating together in phase so that the amplitudes increase. Remove resonance in a room by creating Height X Width X Length ratios that are not multiples of one another (ie 10 X 20 X 30).
Shape - Parallel walls reinforce sound waves, since the angle the sound wave hits the wall equals the angle of reflection and bounces right back to the opposite side. To break up these parallel reflections, design the room with non-right angle corners or rounded walls.
Acoustics - Different surfaces interact with sound waves differently - they may absorb, reflect, diffract, or diffuse the sound. Absorption soaks up the sound, which results in a lifeless, dry sound. Reflection results in a complete bouncing off of the sound from the material, giving reverberation. Diffraction bends the sound around the surface, while diffusion reduces the amplitude of a wave and spreads it in time.
Link to more information on sound diffusion
Link to more information on sound diffraction
The Sound Absorption Coefficient goes from 1.0 for complete absorption to 0.0 for complete reflection. Different materials are assigned a value in this range based on how they react to sound.
Listen to interesting binaural microphone effects here
Monday, October 13, 2008
What is Sound
What is sound? Sound is the vibrations of molecules in the air as the energy is passed in waves of compression and rarefactions. It can be pictured similar to when a stone is dropped in a lake, and the radiation of the waves from that point.
The five main components of sound are: Frequency, Amplitude, Velocity, Wavelength, and Phase.
Frequency is the number of cycles a wave completes in one second, and is measured in Hertz (Hz). For example, if 500 cycles per second are completed, the frequency is 500Hz.
Pitch is how humans psychologically perceive frequency.
Amplitude is the wave's maximum height (crest) and minimum (trough). This is perceived as loudness. Loudness is measured in decibels, which is on a logarithmic scale in powers of 10. In order to double loudness, an increase of 6 dB-SPL is needed.
Velocity is the speed of a sound wave. Generally this is 1130 ft/sec at 70 degrees F at sea level through air. Thicker materials slow down the molecules - 4800 ft/sec in water, 11700 ft/sec in wood. Also, as air warms up, the speed of sound increases, and the inverse is also true.
The wavelength is exactly what is sounds like - the distance a sound wave travels to complete one cycle. Longer wavelengths correlate to lower frequencies, while higher frequencies have shorter wavelengths.
Sound movement examples: http://www.fearofphysics.com/Sound/dist.html
Decibel vs. loudness charts: http://www.gcaudio.com/resources/howtos/loudness.html
Reflection examples: http://www.kettering.edu/~drussell/Demos/reflect/reflect.html
Questions:
1. What is timbre? How does it define different instruments?
2. How does the inner ear convert sound waves to electrical pulses for the brain to understand?
The five main components of sound are: Frequency, Amplitude, Velocity, Wavelength, and Phase.
Frequency is the number of cycles a wave completes in one second, and is measured in Hertz (Hz). For example, if 500 cycles per second are completed, the frequency is 500Hz.
Pitch is how humans psychologically perceive frequency.
Amplitude is the wave's maximum height (crest) and minimum (trough). This is perceived as loudness. Loudness is measured in decibels, which is on a logarithmic scale in powers of 10. In order to double loudness, an increase of 6 dB-SPL is needed.
Velocity is the speed of a sound wave. Generally this is 1130 ft/sec at 70 degrees F at sea level through air. Thicker materials slow down the molecules - 4800 ft/sec in water, 11700 ft/sec in wood. Also, as air warms up, the speed of sound increases, and the inverse is also true.
The wavelength is exactly what is sounds like - the distance a sound wave travels to complete one cycle. Longer wavelengths correlate to lower frequencies, while higher frequencies have shorter wavelengths.
Sound movement examples: http://www.fearofphysics.com/Sound/dist.html
Decibel vs. loudness charts: http://www.gcaudio.com/resources/howtos/loudness.html
Reflection examples: http://www.kettering.edu/~drussell/Demos/reflect/reflect.html
Questions:
1. What is timbre? How does it define different instruments?
2. How does the inner ear convert sound waves to electrical pulses for the brain to understand?
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