A2 Physics Revision - Sound and Acoustics PDF

Revision for A2 and A3 **[The physical of sound]** How does sound propagate, and how do we measure sound? - Sound is made by introduce a source that disturbs the air molecules around it, creating fluctuations above and below normal atmosphere pressure - Through compression and rarefa...

Revision for A2 and A3 **[The physical of sound]** How does sound propagate, and how do we measure sound? - Sound is made by introduce a source that disturbs the air molecules around it, creating fluctuations above and below normal atmosphere pressure - Through compression and rarefaction of air molecules, energy is propagated away from the source in a pattern of alternating air pressure we call a Sound Pressure Wave. - The inverse Square Law - States that 'the intensity of sound in a free field is inversely proportional to the square of the distance to the source' - Energy level depreciate - In a free space a sound from a point source propagates uniformly in all directions- radiating much like an expanding sphere - The intensity of sound decreases as the distance increases The decibel - Decibel (dB) -- a unit used to describe the difference if intensity between two levels - Used measure sound pressure levels (dB SPL) or voltage (dBu) - **[NOT]** a measure of volume or loudness - For Decibels to mean anything we apply a concrete value to 0dB and then all other value are in comparison with that - The Decibel Curve -- twice the power would be 3dB louder but to hear a perceived difference would be 10dB louder - The sentive of the ear is not linear but logarithmic - Max senative for quieter sound and Min Senativity for the louder sounds. - The ear operates over the energy range of 10,000,000,000,000: 1 from the threshold of hearing to the threshold of pain - Sound as time-based phenomenon -- amplitude, frequency and phase (time) - Speed of sound (340m meters/p/second) - How do we calculate the wavelength of any given frequency? - Wavelength = Speed of sound (340 meters p/sec) / frequency - **[Auditory perception and psychoacoustics ]** Phase (time different) - Anatomy of human ear and roll in the link in the sound recording chain and reproduction chain (transduction) - Sound arriving at your ears from the right arrives at the left a bit later. There is a delay and the phase or time different between your ears to established directivity. - Long sound waves (low) the phase or time different is only slight but with short sound waves (high) it will be much greater. The human brain finds it hard to detect the direction of low frequencies compared to high frequencies. The pinna, tympanic membrane, ossicles - Ear Drum (tympanic membrane) vibrates in sympathy with sound waves. - This transferred in the inner ear by the Ossicles -the three bones called Hammer (amplify the sound), Anvil (amplify the sound) and Stirrup (communicates the vibration or amplify vibration to the inner ear) - The Ear Drum (tympanic membrane) is like a transducer converting sound into neutral signals. - Variation atmospheric pressure on the front of the ear drum (tymconpanic membrane) will disrupte the action of the ossicles and unless compensated for by equal pressure from rear of the ear drum. Our presaption of sound is distorted so this compensated for by the eustration tube. Human Frequency response (time, pressure, frequency-range) - Inner ear has a long tube filled with fluid rolled up like a shell called the Cochlea - The tube is divided into upper and lower compartments forming chambers - Sound vibrations travel along these compartments for top to bottom and part through thousands of hair cells link to nerves fibre - These hair cells respond to different frequencies and are split into 24 bands of one third octave spacings. - Centre frequencies for these bands start 50Hz for the first band and 13.5kHz for band 24. - Average person under the age 30 frequency is 16Hz to 16kHz or 20Hz to 20kHz **[Acoustics]** **Room modes (3 types)** - **Natural resonate that occur in a space** - **We caculate the frequency at which a sound wave might perfectly fit into a room using modes** - **Three types of modes** - **Axial are the 'worst' -- with the greatest boost of SPL (2 surfaces)** - **Tangential have half the power of Axial Modes (4 surfaces less energy than the axial)** - **Oblique have a quarter of the power of Axial Modes -- not too much a problem unless they combine with another mode (all 6 surfaces)** - **Can be control with acoustic treatment** - **Axial mode analysis -- LxWxH / speed of sound then for 2^nd^ mode 2x and 3^rd^ mode 3x** - **Continue up to 300** - **Repeat for width and height** - **Arrange findings in ascending order and look for coincidences** - **Frequencies should 'ideally' be spaced 10Hz to 13Hz apart** - **This will determine likely problems where correlating frequencies occur** Control room Acoustics - Flat and evenly distributed frequencies response - An accurate stereo image - For accurate monitoring and manipulation of sound - Near-perfect translation of sound with no loss of intelligibility Live Room Acoustic - Functions of a live room - More possibility's for using a room's 'sound' creatively - Reverb can enhance our perception of a sound and benefit instrument - When recording a live performance, the room becomes part of the sound - Different rooms suit different styles of music - Controlling and manipulating reflected sound energy to enhance sound - Provide a flexible 'recording environment' appropriate to the 'recording subject' **Nodes & Anti-nodes -- standing waves** - **Standing Waves -- sound waves that operate between two or more surfaces emphases one frequencies over another** - **Reflected wave will cross the path of the inserda wave** - **Node = Zero Pressure (out-of-phase cancellation) disruptive area loss** - **Anti-Node = high pressure (in-phase combination) boost in energy** **The problem frequency range in studio acoustics** - **Low frequencies above and up to 200 Hz modes response** - **Low frequencies are longer** - Near field and the far field (free filed and reverberant field) - Near field -- direct sound dormmates - Far Field - Free field -- direct sound dormmates but SPL decrease 6dB - Far Field - Reverberate field - reflections Sound pressure levels - Sound pressure levels lose 6dB for every doubling of distances Room interaction (reflection, transmission, absorption) - When a sound wave meets a boundary a portion of its energy is - Reflected back into the room - Absorbed into the material and transformed into heat due to friction - Transmitted to the other side of the material, which becomes the new sound source - The type of material effects how much is reflected / absorbed / transmitted - Transmittion Loss -- energy reflected, absorbed and transmitted - A measurement of the volume difference on either side of a wall or boundary - Heavy materials - sheet metal, brick, concrete and glass have a high transmittion loses - Frequency specific and will depends on the acoustic properties of the boundary itself - Absorption is the most influential factor in correcting room acoustic - A absorption Coefficient is the proportion of acoustic energy not reflected when a wave encounters an object; 1 = total absorption 0= total reflection - Foam and soft pawious materials -- good absorption and have a score close to 1 - Hard not pawious material have a low absorption and have a score close to 0 - Most absorbed energy is converted into heat - Absorption is really all about mass - Reflection - Smooth surface and flat have the same angle of inserdents to the angle of reflection - Convex disperse sound energy - Concave focuses and concentrates the wave front - Corner 90 degrees reflection back in a separate but parallel path - Diffraction -- occurs when same order or less than the wave front propagate of the source - Edge of the obsure produce a new wave front in air - Will offer a degree of separation but not complete isolation - **[Audio Signals and studio interconnectivity ]** Audio signal as electrical current - Alternating Current (AC) -- flows of electrons periodically reverse direction (sine wave) - Direct Current (DC) -- flows of electrons in constant direction - Audio equipment uses a changing voltage to represent the changing air pressure of sound waves - An analogue electrical signal uses transducer (mics) - Voltage swings from + to -- creating an alternating current - The current can move a speaker back and forward -- recreating air pressure waves Based electrical wiring (live/neutral/ground/Fuse) - Live - Neutral - Ground - Fuse Units of measurement (Volts, Amps Ohms) - Voltage is the electrical force that move current through a conductor - Also, Potential different two ends of a circuit will have a difference in potential different energy - Voltage is measured in V = Volts Measured relative to 'Earth' at 0 Volts - Current is a flow of electric charge in a conductor - The charge is carried by electrons moving in unison - Current ('I' -- from Intensity) is measured in Amps: I = Current A = Ampere (unit of measurement) - Resistance in electricity is something that 'resists' the flow of current - Low resistance = more flow (e.g. good conductors, thick cables, short distances) - High resistance = less flow (e.g. poor conductors, thin cables, long distances) - - - Ohm's Law - V=IR - Impedances matching (guitar high impedances, mic low impedances) - Low impedance - 1-1k Ohms - High impedance -- high than 1k ohms - Guitar amp = high impedance and connected unbalanced 'high z' cable - Microphone = 'Low' impedance and connected balanced cable - Z= impedance ohms unit of measurement - DI -- drop the impedance and changes from a unbalance to balanced - Balanced v un-balanced cables (+ and -- and ground wire) - Unbalanced Cable -- uses screen as its second conductor and this compremises the screen effevtiveness -- will generally result in noise - The longer the unbalanced lead the greater the diatance and potential exposeure to interference - two conductors to translate any signal into a voltage - ¼" jack -- single wire which carries the signal and the screen doubling as a conductor fixed at 0 volts (ground and earth). Unbalanced Jacks TS - Balanced Cable - Minimise noise interference by using two twisted cores - The signals sent out of phase with one another then re-inverted at the destination. Any interference picked up is also inverted and put out of phase. - Common Mode Rejection - XLR -- hot (+ signal) and cold (- signal) wires carry the signal voltage - Common is earth and acts as a shield - Pin wiring -- 1 = earth 2 = hot+ 3 = cold (-) - TRS -- tip, ring and sleeve - Condenser mics require a small power supply to operate - +48 volts supply generated from mixing console or power supply and via mic cable - The supply must free mains noise to avoid hum and must be a constant +48 volts - Always cut speakers turn on 48v on - Patch-bays (**half-normalled (input breaks connection)** and normalled and open) normalled permeant connection - Upper row = output - Lower row = Input - **[Reverb ]** - Reverb measurement (RT60) -- how do we measure RT60? - RT60 -- reverb decay time. The length of time it takes the persisting sound of reverb/reflections to decrease by 60dB below the original impulse level - Reverb properties and impulse-response (early reflectios, late reflections, diffuse, decay and pre-delay) - Early Reflection -- the initial reflection that have bounced off surrounding walls; they reveal the spatial characteristic of the room - Pre-delay - time between the direct signal and the first early reflection - Diffuse reflection - reflection sound waves continue to bounce between multiple surfaces, eventually reaching our ears. Continue stream of late reflections.. - Room interaction (reflection, transmission, absorption) - Digital reverb - Early digital reverb - Convolution reverb -- Impulase responses - - Delay/echo and other time-decay modulation effect - Bucket bridge delays (BBD) -- a serie of storage capacitors carry the audio signal in sequence - Classic tape echo machines -- the record head records a signal to tape, the signal passes the playback head is heard again, the tape goes full circle is over-written at the erase head , recording a new signal at the record head and so on, feedback - Ditigal delay - Flanging -- comb filter - Phasing -- notch filter - Chorus -- **[Microphone ]** - Directional capacity - Directivity how response it is to sound at 360 degree and on a hortonal plane - Response to front and rear (on axis) than the side (off-axis) - Some mics are muilt directional - Cardioid -- effetion at recording in focus and directional, most responsive to sound at the front of the mic not at all to at 180 degree at 90 and 270 degree the responsive characteristic to attenuate. Good for recording multi-instrument ensemble - Super Cardiod -- more focus than the cardiod but with better side rejection. Futher narrow at 90 and 270 degrees so good to reduce senative to sound at these areas. Increase senativity to signal at 180 degrees. Good for singer in a live setting. Singer has little room to move otherwise will sound for mic. - Bi-directional or 'Figure-of-Eight' -- example of a true pressure gradient system, both side of the mic diamphragm are evenly expose the external sound pressure, front of the mic has positive polarity and the back has negative polarity but sound arriving at the side of the mic (90 and 270) on a horitanl plane will arrive at both side of the diaphragm with equal amplitude but oppositing polarity(phase cancelling) so great side rejection. - Omni -- Senative a 360 degree, usualful for recording room ambience and reflected sound energy, in space stereo pairs and 5. Surround array, can be used in radio broadcasting to capsule round table convosations, can cause phase problem when used with other directional mics - Pressure Zone Mic (Boundary) -- uncommon hemi-spheres response pattern, omni directional but it is interfear with by the boundary rather than a gradient. - Design types (dynamics, condenser/capacitor, ribbon) - Shot Gun Mics -- good example of super-cardiod, interference tube to move the capsule further away from the sound source and for reject side channel info and ambient content, hyer-sentivit response to on-axis sound only especially higher frequencies. - Dynamic mic -- afford, rough and common, popular with live event and rehearsal spaces, good are response high SPL and physical shock, good for recording drum, guitar amps, and gun shot and explosions, no phantom power, not a senasive as other mics due to heavy diamphram and coil are slow at reacting to faster, higher sound waves,peak at 5kHz and roll off at 8kHz, can produce colour sound on vocals due to eletro-magnetic induction - Condenser Mic -- phantom power, thin membrane is very light so more senetive, not attached to a coil -- restricted by size, diaphragm is mounted to pick up faster vibration, cannot tolerate for high SPL recordings, great at recording vocal and acoustic instrument, wider and flatter frequency response, impressive response around 12-22kHz, handled careful not as sturdy as dynamic mics, handling noise - Ribbon mic -- oldest mic type, has a extremely thin strip of metal, use electromagnetic induction, most are bi-directional, warm and colourful sounding, wider and flatter frequeny response than condenser mics, damage easily, cannot tolerate high SPL, need a power supply, very natural sounding recordings, good for recording acoustic instruments like strings - Transduction process for above design types? - Electro-magnetic Induction (Dynamic Mics) -- use a system called the moving coil system, acoustic energy in to electric charge, wire coil suspended over a magnet, attached to the coil is the diaphragm which response to sound waves, coil moves back and forwards over magnet inducing alternating current into wires producing an output signal, - Capacitance (condenser/capacitor mics) -- ability to hold charge, diaphragm sit in front a electric charge backplate (48v), as the diaphragm move by the sound sound waves, the capacitors or charge will between the diaphragm and backplates change = amount of electric charge stored, the charging store of energy is proportional to the incoming sound waves and the output is an alternating current, the signal pass through an amplification circuit, - Electro-magnetic induction (ribbon mic) -- a long fine conductive strip of metal (pleated for spring and ridge) is place between north and south pole of a magnet, the ribbon vibrates and induces an electric charge, the electrical output is small so a transformer steps up the output therefore a power supply is needed - Proximity effect what is it? - Performance characteristics in relation to design types - **Pressure and pressure gradient systems** - **Omni pressure mic** - **Pressure and pressure gradient astablish there directivity** - Microphone application and placement **[Signal Path:]** - Pre and Post fader sends -- what are the implications for setting up a cue mix? - Pre fader for cue mix to be independent of our main mix - Post fader for cue mix will affect the main mix remain relative if we turn our fader up or down - Pre and Post fader sends -- what are the implications for FX routing? - Pre-fader aux -- signal is taken before fader so that level is independent - Typically used for headphone mixes - Post Fader Aux -- signal is taken after fader so that level being sent to aux is relative - Typically used for reverb -- we want the amount of reverb to remain relative if we turn our fader up or down - Difference between signal routing on the channel path (record path) and monitor path (return path) - Channel Path -- - Source - Mic 1k-2k ohms or line 10k ohms (don't need as much amplificon) - Input - - processing - add processing you will be committed to the recording (usually bypassed unless turned on) - fader -- control the amount of level/signal being send to a recorder - pan -- associated with the routing matrix - routing -- feed to multi track record media via the routing matrixes - To Multitrack - Monitor Path -- - source Multitrack return -- from an audio interface DAW return - Input - Processing (usually bypassed unless turned on) -- more aggressive use of processing because will not be committed to the recording - Fader - control the amount of level/signal being send to the stereo mix bus - Pan -- move signal in a to stereo or surround field for monitoring - Routing - - To stereo mix - Channel path add EQ or other effect you will be committed to the recording - Split consoles, in-line consoles -- what's the difference? - Split consoles -- two modules or channel configurations one for each path - One side of the desk uses input modules foe the channel path - The other side uses monitor modules for the monitoring path - Not is common now until in an old vintage's studios - In-line consoles -- can accommodate both channel and monitoring path on the same channel strip - input stage you can switch between the mic, line or multi track return and route to either the stereo mix, multi-track inputs or both - saves space and allows for twice the amount of both channel and monitor signal paths - more compact and ergonomic - two sets of faders -- large and small (these can be flipped between channel and monitor paths) - Inserts, auxiliaries, the routing matrix and the stereo-bus (mix L&R) - Inserts -- signal leaves the console to be processed and usually returns down the same channel path. - Used for dynamics and outboard EQ - Auxiliaries -- Sends a duplicate of the signal to be processed aand returns to a different channel strip on the console. - Used for reverbs and time-based processing. - Multiple sources can be sent to the same place - The routing Matrix - The Stereo bus (mix L&R) - Mixing console -- most basic function is to convert several input signals into single output signal; and that's your mix - Core functions; - Amplifier (amplifies audio signals), - routing matrix (direct audio signals to recording devices, processors headphones, speaker) - Stereo/ surround imaging (panning) - Signal processor (signal filtering, equlisation, compressor) - Record stage -- allows us to take signals from a source and route them to recorder - Mix Stage -- allow us to combine signals and sum them to a smaller numc=ber of channels (usually a stereo pair) - 'Confidence Monitoring' -- most of the time we would want to perform both of these tasks (Record and Playback) simultaneously. **[Recording Formants, Multitrack Recording & Digital Audio:]** - Sonic attributes associated with analogue tape recording - Tape Width -- influence Signal to Noise Ratio - Signal to noise ratio can be improved by 3dB by each doubling of track width - ¼ and ½ inch tape - 8 tracks - 1-inch tape -- 16 tracks - 2-inch tape -- 24 tracks - Headblocks - Tape Speed -- with influence the frequencies response of the recording - 50-80Hz low lifted on tape - Concpets and terms associated with analogue tape recording - Wow and Flutter are irregularity with the tapes machine transport (capstan) - Wow -- slow dragging effect - flutter -- fast audibility wobbles in the playback - flux -- magnetics the oxide on the tape - bias - - hysteresis -- the process that causes tape to remember flux patterns - Define the concept of multitrack recording? - The fundamental concept of multi-tracking is having the ability to record multiple and independent tracks of audio in synchronicity. - Method of sound record allows for the separate record of multi sound to create a montage. - Most common method of recording - Les Paul - Concepts and terms associated with digital recording - Sampling -- - 44.1kHz -CD audio - 48kHz -- video-audio and industry standard for most domestic music production - 88.2kHz superior HD resolution to 44.1 and 48kHz, not widely used - 96kHz -- professional studio, require higher processing and storage required - 192kHz - professional studio, require higher processing and storage required - PCM -- Pulse-Code modulation is the process used to translate/convert analogue signal into digital information. - Music Production or Recording the analogue signal is a variable audio signal - The amplitude of the audio signal is sampled at regular points in time to create short samples pulse and each is then quantized to the nearest predefined value. - A/D &D/A conversion -- - A/D conversion -- Analogue-to-Digital converter is the electronic circuit that converts continuous signals to discrete digital numbers. - D/A conversion - Digital-to-Analogue conversion is the reverse - Audio interfaces is an Analog to digital convertor (ADC) - Linear & non-linear formats (tape v random access HD recording, storage and retrieval) - HD recording - Hard Disc Drive (HDD) -- storage requirement -- 1 min of stereo 16/44.1 = 10MB - Performance of HD depends on physical restriction of disk rotation speed: 7200 rpm recommended minimum for recording. - Solid Sate Drive (SSD) - No moving parts = no mechanicals restriction on access times - Test have revealed it to be very fast for recording - Increasingly cheap storage available Difference between analogue and digital recording - Analogue -- - audio is recorded directly onto the storage medium (magnetic tape) - sound is recorded by converting variable SPL's into varying electrical voltage for storage as a phhsical, varying pattern on tape, optical film or vinyl. - Digital -- - audio is converted to a stream of whole numbers first, which are then recorded onto the storage medium (Hard Drive) - sound is also record by converting variable SPL's into a varying electrical voltage, although the storage and reproduction of that sound utilizes a very different process; the varying electrical charge produced from a microphone must first be converted into a series of binary numbers before it can be stored/recorded. **[Equalisation: ]** - Filter-types (Hi-shelf, lo-pass, hi-pass, peaking bell curves) - Low/High Pass -- Attenuate or Boost low and high frequencies at a specific frequencies point - Shelving Filter - - Peaking Fliter/Bandpass Filter - Fully-parametric v semi-parametric EQ - Fully parametric EQ - full control of the full frequencies range from low to high - A fully adjustable Q factor for each frequency - Semi-parametric EQ - Only mid-range frequencies will have adjustable Q factor for bandwidth controls - Low and high frequencies are fixed or predefined shelving - Q (bandwidth), Peak/Notch (boost/cut), Frequency select - Review Fletcher Munson curve in light of equalisation - Human hearing is most sensitive to frequencies around 1-4kHz - Less sensitive to lower frequencies - Human hear distort the perceived balance of frequencies - Small changes in volume can affect the way we hear frequencies **[Dynamic Control System]** - Refering to the dynamic range, or dynamic transient of an audio signal, we're refering to the loud and quiet parts of the waveform - Compressor -- limit the dynamic of a signal, limiting peaks in amplitude allowing quieter parts heard more clear - Limiters - limit the dynamic of a signal, compressor with very high ratio (10:1 / 20:1 or more) - Noise Gates - limit the dynamic of a signal, noise reduction below a certain level, - Expanders - limit the dynamic of a signal, noise reduction below a certain level - De-essers and multiband compressor - limit the dynamic of a signal, frequencies dependant - Define parameters: - Threshold - set the level it will begin to act on a signal - Gain-reduction -- amount turned down by - Ratio -- how much it is turned down - Attack -- amount of time taken for it to reach a specified amount of GR once a signal has risen above the threshold. - Release -- the speed it returns the signal to its original unaffected state - Hard knee -- only react to a signal once it has risen above the specified threshold - Soft knee -- gradually begin compressing a signal