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Setting Up DSP Processors Gordon Moore CTS [email protected] 1-800-821-1121

Setting Up DSP Processors

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Setting Up DSP Processors. Gordon Moore CTS [email protected] 1-800-821-1121. DSP . Digital Signal Processors Can be any device that modifies a digital signal (video, audio – anything) In this class – we will be talking about Audio NOT a “How-to” for specific manufacturers - PowerPoint PPT Presentation

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Page 1: Setting Up DSP Processors

Setting Up DSP Processors

• Gordon Moore CTS• [email protected]• 1-800-821-1121

Page 2: Setting Up DSP Processors

DSP

Digital Signal Processors Can be any device that modifies a digital signal (video,

audio – anything) In this class – we will be talking about Audio NOT a “How-to” for specific manufacturers

Use their training

Page 3: Setting Up DSP Processors

Apologies to any Manufacturers whose screens are not featured – no slight intended.

The CONCEPTS are the core of this course - no brands

endorsed or rejected.

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What to expect Signal pathway organization Setting the array of “modules” or functions available

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Why? Most DSP processors are barely used to their best capability. Units returned for service usually have just the simplest functions

enabled while many enhancements are not touched.

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Most commonly unused or forgotten Compressors Limiters Input filtering NOM bus Delays

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Most commonly set up functions Input Gain (But often badly set) Routing (inputs to outputs) Equalization for outputs Controls interfaces

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DSP organization

Flexible Architecture Fixed Architecture

Dedicated function (one capability only) Multi-function with fixed pathway

Hybrid Architecture – some fixed, some flexibility in routing

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Flexible ArchitectureAll functions can be configured in a “drag-n-drop”

environment. Audio functions can be placed almost anywhere along the signal chain in any order.

Characterized by a drag and drop GUI (Grpahical User Interface) and/or “fuel gauge”.

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Sample

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Advantages

Complete flexibility – you can do some amazing things within a single box and develop very complex signal paths. Excellent choice for systems where complexity and/or multiple applications may come into play.AirportsLarge scale paging systemsComplex communications systems

Very little you cannot accomplish.

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Disadvantages

Flexible Architecture may be more costly on a per channel basis Requires more DSP power – memory register stacks must be

allocated for any eventuality – code space cannot be optimized.

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Fixed Architecture Dedicated function

Does one type of function Compression/limiting Or Equalization Or Signal Routing

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AdvantageFixed architecture is simple to set up and operate – may not

even require a computerLess cost for the box – may not be as cost effective as a

combined DSP capabilitySets up much like analog counterpartExcellent choice for existing system upgrade

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Disadvantage Very limited in scope of function May not be very scalable

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Fixed Architecture

Multi-functionHas multiple functions in a fixed pathwayGenerally fairly cost effectiveLimited in terms of routing and or set up choices.

Advantage – predictable known good pathwayNo gauge

Disadvantage – few if any routing choices

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Hybrid Architecture Combination of routing choices plus fixed multiple function signal

pathway Advantages –

Allows flexibility as far a signal routing goes – what inputs show up at what outputs Optimizes DSP processing power – memory stacks and registers can be more

tightly packed Disadvantages

May not fulfill all needs in a system

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Hybrid Architecture

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Enough Boring stuff – Let’s set up some functions Input Gain Filters – Input and/or Output

Equalization Feedback suppression Crossovers Noise reduction

Dynamics Compressors Limiters Gates

Routing

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Dynamics Those functions affecting gain structure and levels

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INPUT GAIN Most important setting – GET THIS RIGHT! Always set up – but not necessarily well set up Microphones

Handheld Vocals = 35dB minimum Handheld Presentation = 45dB Gooseneck desk = 45dB Boundary mic = 55dB Any further away = 60db+

Ceiling – as hot as you can get it

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INPUT GAIN Multimedia

Unbalanced? Consumer = +10 Balanced? Professional = 0 to -8

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Signal to noise ratio

Mic Preamp Mixer Signal processors Amps

LineLevel

Mic Level

Noise Floor

Poor S/NRatio

Page 25: Setting Up DSP Processors

Signal to noise ratio

Mic Preamp Mixer Signal processors Amps

LineLevel

Mic Level

Noise Floor

GoodS/NRatio

Page 26: Setting Up DSP Processors

Signal to noise ratio

Mic Preamp Mixer Signal processors Amps

LineLevel

Mic Level

Noise Floor

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THRESHOLDS The level at which the desired function becomes active

Generally speaking a lower threshold level means it will activate earlier.

Recommended starting threshold for most line level (post preamp) functions = 0dBu

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AGC and/or Levellers Automatic Gain Control RAISES gain if signal too low Compresses if signal too high BE VERY CAREFUL with these

Can run a room into feedback if used on amplified inputs Primary application – to capture weak signals for recording or transmission Start with threshold set at 0dB – keep gain centered at line level

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Ambient Level ControlUses a reference microphone to measure room noise level the

automatically adjusts system for noisier environment

Reference microphone may be dedicated microphone – only purpose is reference, or may be designated microphone – used in system but designated to be the reference signal

Read the manual

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Page 31: Setting Up DSP Processors

COMPRESSORS Control dynamics - loudest to softest Useful for keeping level under control Meek versus motivational speaker Rarely set up

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Compressor settings - Ratio

Ratio – The amount of actual level increase above threshold that will yield ONE decibel in actual gain change after the compressor. Example – 3:1 ratio

For every 3DB the gain increases above threshold, the final level will change only one dB

SO, if level jumps 9DB, the final level will jump only 3dB FM is broadcast at typical 10:1 ratio

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Compressor settings – Threshold, Attack, Release

Threshold - Level at which compressor begins to engage and affect level.

Attack – time in milliseconds the compressor begins to make changes after level exceeds threshold

Release – time in milliseconds the compressor lets go after level settles below threshold.

Makeup or post compressor gain Compensation in level to make up for compressor reduction in signal.

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Suggested setting for compressors Speech systems (conference rooms, boardrooms, etc)

Ratio = 3:1Attack = 10-20msRelease = 200-500msThreshold = 0If initial input gain was set to result in 0dB level, then it would take a 60dB

increase at the mic to hit the +20dB limit of input (clipping)MUSIC or Multimedia – try increasing ratio to 6:1 BE CAREFUL – too much = bad

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LIMITERS Basically, a compressor with an infinite ratio Absolute ceiling to maximum level Protects downstream gear by preventing severe clipping and

overdriving amps and speakers Many amplifiers have built in limiters to protect themselves. ALWAYS set limiter threshold above threshold of compressor

Otherwise, compressor will never engage Good for spikes like dropped microphones, cymbals, plosives

(P,D,T)

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Suggested Limiter settings Threshold –

15db higher than compressor = +15dB if 0dB for compressor Attack – faster than compressor = 2ms or quicker Release – 200ms or less

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Some dynamics filters can be frequency specific Compressor with low pass filter good for controlling proximity effect

while allowing high frequencies to pass unaffected.

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EXPANDERS

Increases gain if signal very low such as weak talker BE VERY CAREFUL HERE – Expanders, in an

amplified environment, can push system into ear bleeding feedback.

Primarily intended for recording and or transmission.

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GATES Gates activate a channel, allowing it to pass, once the

level passes above the threshold. Found in some automixers Useful for noise control (noisy multimedia source, for

example) Originated in music

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Attack , Release, Threshold Attack – try relatively fast settings, 1ms to begin Release – start at 50ms Threshold – depends on place in chain

If after input gain – 0dB is a good starting place Lower if not getting a reliable start

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Noise Gate

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Spectral Affects the frequency response of the system

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EQUALIZATION (Inputs) Equalization is one of the most commonly used functions

Input EQ is generally for tonality control – adjusting the tonal content so each input sounds similar.

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EQUALIZATION (Outputs) Generally used for speaker compensation Adjusting for “quirks” or characteristics in the loudspeaker response. You cannot EQ a “room”

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Equalization patterns Pass – Low, High or Band Shelving – Low or High Parametric - Notch Graphic

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“Q” No, not James Bond’s gadget guy The ratio of filter width to depth at 3dB roll off points

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Low Q – Wide band width

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High Q – Narrow Band width

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Filter Slope or “Order” Rate of attenuation on filter - Shown in terms of dB/octave

Octave – doubling of frequency First order = 6dB per octave Second Order = 12dB per Octave Third Order = 18dB/Octave Fourth Order – 24dB/Octave Each order equals another 6db of roll off.

That means 4 times factor in power level If 6 db down from 100watts = 25 watts.

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First Order – 6dB per octave (High pass)

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Same filter – Second order

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Eighth Order- 48 dB/octave

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Parametric Fully configurable Boost or Cut adjustable Center Frequency selectable Q selectable

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Parametric

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Band Pass

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Low Pass

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High Pass

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Shelving – Boost or cut, High or Low

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Filters - Graphic EQ

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CROSSOVERS

Used for bi-amplified and tri-amplified systems Low frequency content sent to bass amplifiers

Mid and high range sent to appropriate amplifiers Separate amplifiers involved Large scale concert systems High order - 4th to 8th order filters Bass – 250Hz or lower For tri-amplified – start at 4K for high pass

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Page 62: Setting Up DSP Processors

FEEDBACK SUPPRESSION Should be set up LAST after Equalization Smooth response FIRST, then take care of Feedback

nodes A time domain issue coupled with Frequency domain Notch Filters – very tight But too many can badly affect content

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Set up tip – One mic at a time 1. Before final EQ – run feedback “eliminator” first 2. Make note of first three feedback freqs 3. Construct three very tight notch filters at INPUT on those

frequencies 4. Reset the feedback filters 5. engage feedback filters again after equalization and system is

at operational levels.

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NOISE REDUCTION

Popular new algorithms that “sample” the noise floor Noise floor – acoustical and electronic noise that is NOT wanted in system Air Conditioning Fan noise

Laptops near boundary mics computers Projectors

Electronic noise – noisy sound cards in computers etc.

Page 65: Setting Up DSP Processors

Setting Noise Reduction Filters

Canceller Depth – depends on the amount of noise Quiet conference room with little to no noise may not need this. Computer and projector fan noise – try starting at 9dB. Heavy room noise – large attendance training room or bad air conditioning

rumble, try 12dB Remember, these filters remove spectral content – none are perfect and

they will all affect your room response. DON”T GET CARRIED AWAY!

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Page 67: Setting Up DSP Processors

Signal Generators White noise – equal energy per frequency Pink Noise – equal energy per octave Tones Primary use – test and measurement

Use pink noise to set up your levels in the room If your gain structure is correct – this will allow amplifiers settings tobe accurately set up Use test tones for gain staging and/or speaker alignment

Secondary Use Noise masking – covering conversation or background ambient noise Alarms

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ROUTING Matrix –

Rarely overlooked Determines which inputs go to which outputs

(technically speaking “gozindas to gozoudas”) Some traps here Watch for

Feedback loops NOM bus assignment – VERY IMPORTANT

Page 69: Setting Up DSP Processors

Selecting NOM bus and action NOM= Number of Open Microphones Determines interactions of microphones in automixing Failing to select correct NOM interaction can affect echo cancellation,

and gain before feedback Choices may include

Chairman Over-ride Auto mix (or Normal) Background (or Ducking)

Page 70: Setting Up DSP Processors

Feedback loops

Common error in setup – requires careful documentation and double checking – especially when dealing with mix minus conferencing systems.

Sending an input BACK to itself.

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Delays

Often neglected Primarily used for time alignment

Loudspeaker stacks in large venues Input alignment Loudspeaker alignment

Secondary use for spatial referencing (Haas effect) Localization of sound based on first heard

Set up tip – Try setting for 10ms or 10ft or 3.3m FARTHER than distance would indicate. Example – Speaker is 68 ft from stage – instead of 60.5ms, try 70.5.

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Some architectural Do’s and Don’ts In Flexible Architecture – General rules – always exceptions

Do NOT put compressors before Equalizers/Filters The EQ can take out energy that would falsely trigger compressor – so put

compressor AFTER filtering Do NOT put limiters before compressors - Do NOT put limiter thresholds lower than other devices – especially

compressors Be careful about thresholds for automixers – (gated designs)

Too low and room noise will open mics Too high and you may lose the first characters Shared gain mixers will not have thresholds

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Hybrid Architecture Do’s and Don’t Don’t create a feedback loop – be careful where you send inputs Don’t go overboard on settings

“Too much of a good thing is wonderful” does NOT apply here.

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Troubleshooting tips Ringing or severe echo in audio

Internal feedback loop Signal generators or incoming conferencing too loud

Poor input gain structure – HOT IT UP! HISSssssssssss or noisssssssse in system

Bad gain structure – inputs too low – Amplifiers too high University reasoning – wrong solution to the right problem.

Amps set to full output Inputs set low Correct solution – lock up the amplifier controls.

Page 75: Setting Up DSP Processors

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