• Audio Signal Processing Basics ISO87 Course Description

• Get the basics of compressors, delays, automatic gain controls, noise cancellation, automatic mixers, gates, and acoustic echo cancellation in conferencing and meeting room audio applications. • This course covers the configuration of audio equipment.

2 Listen – What problems have you heard?

What problems can be solved with DSP?

3 Problems

• Audio levels too high to low or inconsistent • Reverberation • Echo • Distortion • Noise Proper System Design

• Gain before feedback • Direct to Reflection ratios • Signal to Noise ratios • Other poor performing equipment in the system

5 Sound Level

6 Word Articulation

7 Setting Levels to Optimize System Performance

• Know how the equipment metering works • Know the level that clipping happens

8 dB Units

• dB - logarithmic ratio of two values • dBu - for measuring voltage – 0dBu = 0.775 volts • dBV - for measuring voltage – 0dBV = 1.0 volt • dBFS - decibels relative to Full Scale – Refers to the maximum signal level possible Gain Structure

To high - Clipping Maximum Level (dBFS) That’s the way we like it

To low – Poor S/N Noise Floor

10 Distortion

• The shape of the original sound wave is modified • It occurs in the audio circuitry, algorithms, and encoding • Clipping the signal causes harmonic distortion Bit Resolution

• Every bit gives you 6 dB of dynamic range • The bit resolution sets the total dynamic range available • CD’s are 16 bits while DVD’s go up to 24 bits • A CD has 6*16 = 96 dB of dynamic range • A DVD has 6*24 = 144 dB of dynamic range

• When we do not send enough signal level to the A/D converter it is like lowering the systems bit resolution Input Level Adjustment

Analog A/D Average and Peak Levels

14 • Analog vs Digital Gain Adjustments

Analog A/D Digital D/A Analog Input Converter Processing Converter Output

All Level adjustment are Digital Headroom

• Why 0dBu or +4dBu? – Most systems need “headroom” – generally 20dB will handle anything – Most professional gear can handle up to 20 or 24dBu – Capping “normal” level at 0 allows up to 20dB of extra loud signal before distortion or clipping Mic Distance

• Inverse Square Law – Each doubling/halfing changes level by 6dB • If level at 1 foot (.3m) is 74dB-SPL then – At 6” = 80dB-SPL – At 3” = 86dB-SPL – At 1-1/2” = 92dB-SPL • A total difference of 18dB • Set the analog input gain adjustment with the maximum level that will be used. Meter Types and Scales on Digital Processors

• Meters which monitor audio levels are typically one of two varieties: RMS,VU (Volume Unit) or PPM (Peak Program Meters).

19 Meter Example

• Peak Meter • Adjustable Reference Level

20 Meter Example

• Peak Meter • We do not know the attach and release speeds • 0dBFS (full scale) • What level dBFS is indicated on the meter

21 Meter Example

• RMS meter w/ clip indicator • Level in dBu – shows max of 20dBu • We do not know the attach and release speeds

22 Comparing Meters

Peak w/ Reference dBFS Point at 0dBu Level Dynamics

• Automatic Mixer • Compressors • Limiters • Automatic Gain Controls/Levelers • Gates

24 Automatic Mixer

• Gating • Gain Sharing

25 Automatic Mixer

• High quality audio becomes more difficult to achieve as the number of open microphones increases. whenever multiple open microphones are used. These problems are:

– Build-up of background noise and reverberation – Reduced gain before feedback – Comb filtering – Increased Echo when conferencing Noise

Minimum desired S/N ratio Noise

S/N ratio with eight S/N ratio with one Microphones open Microphone open Gating Auto Mixer • Gating Gain Sharing Automatic Mixer

• Sums all of the inputs, measures the level of that sum, and attenuates all inputs by the difference between their level and the level of the sum. • Basically automatically riding the channel levels, turning up microphones in use, and attenuating unused microphones. • No gate thresholds that may trip erroneously • Eliminates gate timing issues

32 Which to use

• Personal Preference • Gain Sharing may be more natural • Gating allows more control of mic activation – Better for acoustically bad rooms are microphones at a distance – Better for conferencing applications Compressor

34 Dynamic Range

35 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, output or post compressor gain – Compensation in level to make up for compressor reduction in signal. Limiter

37 Suggested Setting for Compressors

• Speech systems (conference rooms, boardrooms, etc) Ratio = 3:1 Attack = 10-20ms Release = 200-500ms Threshold = 0 Common Application

• Microphones where the distance to talker ratio can change by a factor of two or more • Telephone line that may connect to a conference bridge.

39 Automatic Gain Control

40 AGC and/or Levelers

• Automatic Gain Control • Raises gain if signal too low • Reduces it if signal too high • BE CAREFUL with these – Can run a room into feedback if used on amplified inputs Common Application

• Inputs from program sources that the user can adjust. • Telephone line that may connect to a conference bridge. • Microphones where the distance to talker ration can change by a factor of two or more

42 Noise

• 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. Noise

• Band limiting with filters • Active noise Reduction • Automatic mixer

44 How is Acoustic Noise Removed?

• The best method is to get rid of the thing that is creating the noise • Or move the mic away from the noise • Complex DSP algorithms are used to reduce the level of steady-state background noise • These are called noise suppressors or noise cancellers Noise Cancellation

• If the ambient noise is repetitive in nature (mechanical, etc.) Active noise cancellation may be used to increase the S/N ratio and allow the microphone to talker distance to be increased. • For each 6dB of noise cancellation the distance can be doubled Noise Gate/Downward Expander Transient Noise Reduction Echo

• Conferencing – Acoustic Echo Cancellation – Automatic Mixer • Sound Reinforcement – Low latency in audio equipment and network

49 Echo Cancellation

• Why an Echo Canceller? How it Works

• A Room is a Filter!

INCOMING SIGNAL CHANGED SIGNAL

Room Acoustic Model of the Room Response

Filtering of the incoming signal is INCOMING SIGNAL Filter CHANGED SIGNAL performed by combining an

Acoustic model of the CHANGED SIGNAL INCOMING SIGNAL Room Filter Model room response with the signal using a process Convolved With called convolution. Reference (audio to be removed) Echo Canceller subtracts reference from Local Microphone signals (echo) AEC Output Local mics to the far side Local microphone(s) • In some systems Non-linear processing (NLP) is used after the echo canceller to subdue residual echo. • NLP attenuates the residual echo by a variable amount based upon the talk state. The NLP block may also includes a comfort noise generator. Tail Time

55 Successful Setup of AEC

• ERL • Reference • Reference Level

56 ERLE Measurements

57 AEC Setup Tips

1. Start with the power amps turned down all the way. Adjust the mics input gains for about -20 dBFS during normal talking level into the mics. 3. Adjust the rest of the gain structure through the entire system for unity gain 4. Then establish a connection to the far end, and then slowly bring up the level on the amplifiers, until the appropriate loudness is obtained. 6. Check the ERL. ERL will normally be negative; if it is positive or too negative, it may indicate a gain structure problem. 7. Once operational, make minor level changes as required, but do not change the level of the amplifiers 3. Adjust NLP settings If echo is still being heard, switch to higher setting. Reasons for Bad AEC Result

• Bad gain structure • Local reinforcement is too loud • ERL showing more than +0/-10 dB • The reference signal tapped off prior to dynamics, filtering, or delay processing • Signals routed to the AEC reference is not correct. (far side audio only) • Signal feeding the AEC reference not within the recommended range. Equalization

• Generally used for speaker and microphone compensation • Adjusting characteristics in the loudspeaker response. Filters - Parametric

61 High Pass Low Pass Feedback

• Filters – Automatic Feedback control • Automatic Mixer

64 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 filters can badly affect content Setup 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. Can Be Simple Mic Inputs

Mic AEC/NC HP/LP/EQ To Gating Automixer Input /AGC for Conferencing

HP/LP/EQ To Gain Sharing /FBX Automixer for Sound Reinforcement Program Inputs

Program Input or AGC/Compression To Routing Matrix Remote /limiter Mixer Audio Speaker Outputs

HP/LP/EQ/Limiter To Loudspeaker From Routing /Delay/Crossover Output System Testing and Commissioning

• Measure and record ambient noise, A-weighted, slow. Ambient noise levels measured under all HVAC and with other room equipment operating. • Sweep speaker systems with high-level sine wave. Listen for buzzes or rattles. • Measure speaker coverage uniformity. +/- 2dB • Test each microphone’s polarity. • Test each loudspeaker’s polarity. • Test all mics for signal flow. • " EQ filters are adjusted to produce a flat frequency response +/- 2dB over the entire speech range: From 100 Hz to 8kHz, flat within plus or minus 2dB." • Confirm acceptable audio levels to and from the Codec with the system adjusted provide an average output level of 0dBu. • "Calibrate audio system inputs so there is zero or minimal difference between any input signal level. Balance all audio levels in the Audio Mixer so that all sources interface to the internal matrix mixer at 0 dBu. Make adjusts to for PC inputs with the PC set to maximum volume."

71 System Testing and Commissioning

• Verify auto-disconnect on video conference • Verify ring indicator is on phone • Verify that ring indicator is on video conference • Verify that auto-answer is off on phone • Verify that auto-answer is off on video conference • Verify that noise cancellation is active on microphone inputs and on the telephone interface. • Make a test video call, Make audio add-in to video call. Listen to duplex and for echo. • Measure and note ERL, ERLE and TERL for both phone and video conferencing with audio coming from the distant site. System shall perform at nominal operating levels in a full duplex mode without echo or latency. System Testing and Commissioning

• Configuration – Proper system design – Proper gain settings – Configure AEC – Configure other processing – Test in all use states

74 Questions

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