Ten Audio Terms That May Become Obsolete

Ten Audio Terms That May Become Obsolete Thanks to AES67
April 26, 2018

By Anthony P. Kuzub, Ward-Beck Systems

https://daleproaudio.com/blogs/dale-tech-corner/ten-audio-terms-that-may-become-obsolete-thanks-to-aes67

The audio and video industry is experiencing an IP renaissance. With the deployment of AES67 networked audio solutions, certain terms are in need of revisiting. Though these terms are commonly used in analog and digital transmission, with AES67 and SMPTE ST2110-30 the mechanisms of moving audio are completely re-invented. The terms must follow. These obsolete audio terms are discussed with an update offered for its IP equivalent.

Split
A single microphone on stage in a performance has to feed the talent, the audience, the recorders, and the broadcaster. We’ve been relying on iron core isolation transformers with miles of fine copper to perform this splitting of delicate signals, for decades. Lately, thesesame electrical primitives in use between the front end of multiple digital mixing consoles. Different sample rates, different clocks, and vendor proprietary ecosystems force this. The redundancy of analog to digital conversion with activeand passive splits is no longer needed on a converged network. With AES67 being multicasted onto the network an engineer subscribes to the sources needed to perform the task of signal splitting. The network switch is the most powerful audio splitter ever made and costs less than the road-case for the splitter one can pay thousands for. It’s time to stop splitting and start multicasting.

Patch
We have been weaving, making and breaking audio signals patches since we’ve had a need for them. As the quality of microphones went down, the quantity of them went up. As we made studios and facilities larger, the need for flexibility went up. Multiple sources from different rooms needed to be strung through buildings and hard-wired for a limited and specific purpose. As AES67 is using a layer 3 networking protocol, the need for single application rigid physical infrastructure disappears. As all of these signals are converge to a network, we’re not patching them from source to destination, the destination is joining an IGMP stream. As the sources populate the network, the receivers can have access to that signal. We’re not using patch cables anymore to connect sources and destinations. We’re trading a Session Description Protocol (SDP) and we’re using software to request a stream delivered through IGMP. We’re not patching, we’re joining IGMP streams!


Distribute
Traditionally we push audio to a distant load through a transmission line. Signals on a network do not need to be distributed as they were in the past. With AES67 and SMPTE ST2110-30, you are subscribing to the stream and the networks flow control, Internet Group Management Protocol (IGMP), handles the join connections for you. The classic audio distribution amplifier is over 50 years old, 1 input 8 outputs. 1 transmitter to 8 fixed receivers. With AES67/SMPTE ST2110-30 you have 1 transmitter and (n) receivers. The need to distribute signals is no longer required as they are available anywhere and everywhere at the same time. AES67 signals are Discovered using a mechanism like Ravenna (MDNS-Bonjour), Session Announcement Protocol (SAP-Dante Controller) and Session Initiation Protocol (SIP). These discovery mechanisms are a byproduct of telco-industry.


System Ground
When the front of house mixing console, a hundred plus feet away from the stage, is receiving microphone signals it is very susceptible to noise. If the ground at the console has a different potential for grounding than that of the stage you’re in trouble. Ditto TV studio, Radio studio, Recording studio, Mobile truck etc. With AES67 and networking technologies there is no need for a common system ground. There is no benefit to a common system ground because your sound that’s digitally encoded is now packetized data. The sound is not susceptible to external environmental interference within the transmission path. You cannot bond fiber optic glass to earth and optical pulses are not susceptible to electromagnetic interference. And, ethernet cables are transformer isolated from the electrical system they are operated on. Power over ethernet derives its own reference relative to the switch that generates it. Everything is floating relative to everything else!

Word Clock
Word clock is an imperfect one-way communication system for synchronization. Because of its unidirectional transmission, a receiver of word clock cannot know how far off it is from the generator. Word clock is a one-way conversation with no start, no finish and therefore no linear reference. With AES67 and ST2110-30, audio is synchronized using a precision time protocol (PTP) designed by the IEEE(1588). PTP is a conversation between a master clock server and client edge device. Algorithms are used for elections of the best master clock; this keeps the network in sync and always with a clock. A PTP master clock can be generated by an oscillator locally or slaved. There is no clock source on earth quite like the one not on the earth. Using GPS synchronization, (n) number of stations/remotes or devices can be perfectly synchronized and sync locked continents apart. SMPTE 2059 and PTPV2.1 are the new timepieces for your audio and video networks. What good is word clock when you can be world clocked?

Surround Sound
There is no naturally occurring stereo sound. If that’s true then there is no naturally occurring surround sound. All naturally occurring sound is monophonic, the rest of what we hear must be reflections and influences of the acoustic environment. Mono sounds must be positioned somewhere in a playback environment to give position, depth, width and now (finally) height information relative to the listener’s position. With a traditional five point one audio system we are bussing signals to emulate the above.

With AES70 we can now encode a set of Open Control Architecture parameters to deliver positional information with what could become sample accuracy in a stream of data. Pan, tilt, dolly, roll, truck, pedestal and geo-positional information of a microphone can be captured in the AES70 class architecture. Like an access database, you need a Key, its Length, and the Value. These parameters can then rendered by a multichannel speaker system or encoded relative to Head-related transfer functions in personal headphones. Why be surrounded by sound when you can be immersed in it?


Switching
An audio switch toggles to make a result based on what is feeding it. If you have multiple sources and you need to switch between them, you must receive all the sources and switch between them. Switching between multiple sources requires a lot of wiring fed from distribution systems or large scale matrix systems. With AES67, devices capable of subscribing to any source that is on the network at any time. A source is described in a Session Description Protocol defined by AES67 to include Clock source, Payload format, and stream protocol information. Changing the receivers SDP switches its source. Most devices can be automated and switched remotely with the pushing of a JSON (JavaScript Object Notation) object containing the desired SDP. To ‘switch’ between two sources the receiver changes its join of IGMP streams. Switching is out, choosing the SDP is in!



DeMux – Mux
Somethings Done Iterating. The serialized transmission of multiplexed video is coming to an end. With a converged network we are seeing the things once bundled together be split apart and scattered across the network as unique streams. With SMPTE ST2110 the audio, data, and video are unique multicasts. The best solutions to manages these devices is the NMOS (Networked Media Open Specifications). Its model refers to these elements in a hierarchy of Sources, Flows, and Grains. The audio, data, and video are grains that can be combined to create a flow that becomes a source of the device. The flows from sources are then received and split back into individual grains. Goodbye multiplexing, hello Source. Goodbye de-muxing, hello flow-grains.


 
Input List
Inputs are dynamic. As systems evolve and changes are made on the fly it is very hard to have an input list 100%. Being up to date takes paper and runners; unless, you’re using the same exact spreadsheet. The use of ‘living documents’ such as Google Sheets or Smartsheet allow all operators to have immediate access to the current information, everyone will now be on the same sheet. With AES67, your patching can be automated. With AES67 you are never changing the parameters of the transmitter you are only changing what is being received where. If all of your multicast sources were in a database, with metadata, it could be used to dynamically update your console’s inputs. When you realized that most modern digital consoles are just database driven analog console emulators, you wonder why we’re still passing around paper input lists with numbers instead of a Database of raw accurate data.



Recording
Sound recording is an electrical, mechanical, electronic, or digital inscription of sound waves. In AES67 you have none of the above. Data packets that contain transmissions are moved over a network. Within those packets is a representation of the digital inscription but the packet itself is not the inscription, therefore not a recording. Wireshark could capture the ‘packet storm’ that is on a network. If all of those packets were intercepted captured to a reliable and readable capture format one would accomplish the “recording of AES67”. This process would capture the audio on the network without any introduced signal processing, conversion or dependence on synchronization. The packets captured are individually time-stamped and could be re-assembled for playback. We’re not recording anymore – we’re capturing.

The Scalable 5.1 and Stereo Audio Mixer

The Scalable 5.1 and Stereo Audio Mixer

The 5.1 and Stereo Audio Mixer is a versatile, fully featured Audio Mixer designed for scaling to a wide variety of production needs, including Radio, Television, Podcast, and Web. It is capable of handling complex setups featuring over a hundred Mono, Stereo, and 5.1 inputs and outputs, while also supporting simpler configurations.

Scalability and Configuration

The mixer’s scalability and resource utilization are defined when an instance is launched using its API. Multiple instances of the mixer can operate simultaneously within the same compute server or server cluster.

Individual instances are highly configurable in terms of:

  • Input Channels: Defining the number and types of input channels (mono, stereo, 5.1).
  • Busses and Outputs: Customizing the number, type, and capability of mixing busses and outputs, including Mono, Stereo, 5.1, Program, Aux, Mix-Minus, AFL, PFL, and Subgroups.
  • Processing Functions: Specifying the processing functions included in both input and output chains, such as Gain, Dynamics, EQ, and Delay.

The architecture is designed to separate inputs from the audio mixing functionality, allowing the mixer to integrate with diverse audio sources and destinations.


Input Channel Processing

Each mixer instance can be configured with any number of mono, stereo, and 5.1 input channels. The signal processing pipelines are built using basic processing blocks. These blocks can be configured to operate on mono, stereo, or 5.1 channels where appropriate, and can be bypassed/turned off within a pipeline without impacting processing delay or current settings.

Basic Processing Functions

  • Gain Processing: Used to adjust the level of the incoming signal.
  • Filter: Features user-configurable High-Pass and Low-Pass filters with adjustable frequencies and slopes to eliminate unwanted frequencies.
  • Parametric EQ: A four-band equalizer that allows precise sound shaping through individual adjustment of selected frequencies by applying gain or reduction within a specified bandwidth (Q). Bands can be configured as Bell-shaped, low-shelf, or high-shelf.
  • Delay: Provides user-definable delay, adjustable in 1ms increments.
  • Insert: Required to send and selectively receive an audio signal to an external processing device (e.g., external compression or additional EQ).

Dynamics Processing

The mixer features several dynamics processors, all of which can be controlled by an external audio signal via a Side Chain Input (e.g., for ducking in the compressor or selective un-muting in the Audio Gate).

  • Compressor: Controls dynamic range by adjusting the volume of signals that exceed a set threshold. It offers user-configurable settings for compression ratio, attack, release, knee shape, and makeup gain.
  • Limiter: Caps the maximum signal level using thresholds and response times to prevent distortion from volume peaks.
  • Expander: Increases the dynamic range by attenuating volumes below a set threshold, with adjustable parameters for ratio, attack, release, and range.
  • Audio Gate: A dynamics processor that selectively mutes audio signals falling below a defined threshold, with controllable settings for attack, release, hold, and range.

Stereo-Specific Functions

Available only on stereo inputs, these include:

  • Stereo Control: Offers options for manipulating stereo audio signals, such as 1:1 transparent, Swapped, 1:~1 transparent with one Phase flip, distribution options (1 to 1 and 2, 2 to 1 and 2), Mono Summed (L + R), and Mid-side decoder.
  • Stereo Width Control: Allows adjustment of the perceived width of the stereo field, with expansion, narrowing, and a neutral position.

Channel Level and Auto-Mix

  • Input Fader: An attenuation stage that sets the input level in a mix. A single fader controls both stereo and 5.1 channels.
  • Gradual Transition on Fader: Allows the fader to gradually transition to the desired level, defined by an In Level, Out Level, In Duration (ramp up time), Fade Out Delay, and Out Duration (ramp down time).
  • Input channel VCA: A second attenuation stage controlled by a group fader.

The Auto-Mix processor autonomously adjusts the levels of a group of input channels in real-time based on signal activity and channel priority, using cross-fading for smooth transitions. A minimum of 4 Auto-Mix groups can be configured, and the processor automatically adjusts when channels are added or removed. Configuration parameters include the input channels, a Bypass/Process state, Relative Weight, Attack time (mS), and Release time for each channel.


Input Channel Routing

Positioning and Panning

  • Mono to Stereo Paner: Positions a mono channel across the L and R channels of a stereo signal.
  • Stereo Balancer: Re-positions the L and R channels within a stereo signal.
  • Mono to 5.1 Positioner: Positions a mono channel across a 5.1 surround sound field.
  • Stereo to 5.1 Positioner: Positions the Left and Right channels of a stereo signal across a 5.1 surround sound field.
  • Next Generation Audio (NGA) 7.1.4 Multi-Dimensional Panning: Immersive placement controls allow sounds to be moved in three dimensions across a configuration that extends standard 5.1 by adding four height channels and two side speakers.

Bus and Output Feeds

Each input channel has dedicated selectors and level controls to contribute to different busses, and can feed from various tap-points:

  • Program and Subgroup: Each channel has a dedicated selector to contribute.
  • Mix-Minus: Each channel has a dedicated Mix-Minus exclusion selector and a dedicated level control for each Mix Minus bus contribution. Channels can feed from Pre EQ, Pre Fader, or Post Fader. A Talkback Interrupt function routes a designated talkback input to another channel’s Mix Minus bus.
  • Aux: Each channel has dedicated selectors and level controls to contribute to each Aux Bus, feeding from Pre EQ, Pre Fader, or Post Fader.
  • Direct: Each channel has a Direct out that can feed from Pre EQ, Pre Fader, or Post Fader.
  • PFL (Pre-Fade Listen): Each channel can contribute to the PFL bus from Pre-Fader, Post-EQ / Post dynamics, Pre-EQ, or at input.
  • AFL (After Fader Listen): Each channel can contribute to the AFL bus.

Other Input Functions

  • Alternate Input Source Selection: Typically positioned after the gain stage, this selects an alternate input source while preserving existing channel settings, often used for critical applications like alternate microphones.
  • 5.1 Downmix (Fold Down): Creates a stereo Left Only, Right Only Down Mix (Lo/Ro Fold down) from a 5.1 surround sound channel. The processing uses the formulae: and , with LFE Muted.

Busses and Outputs

Busses are internal pathways for routing and combining multiple channels. The mixer can be configured to include multiple instances of each bus type.

Program Busses (Main or Master Busses)

These form the main program output. The mixer can be configured with one or more independent Program Buses.

  • They feature a level adjustment immediately after the summing block; a single adjustment controls all channels in a stereo or 5.1 Program.
  • Each bus can be configured to include a Compressor, Limiter, and Delay.
  • Supports NGA 7.1.4 Multi-dimensional audio.
  • For parallel 5.1 and stereo productions, the mixer maintains parallel 5.1 and stereo program buses, each with its own output, or provides a 5.1 program bus that is downmixed at the output.

Subgroup Busses

Used to group multiple sources for common actions. The mixer can be configured with one or more independent Subgroup Busses, which can be Mono, Stereo, or 5.1.

  • Each Subgroup Bus includes a level adjustment after the summing block.
  • They can be configured to include a Compressor, Limiter, and Delay.
  • Subgroup buses can be re-entered as an input to other buses.

Mix-Minus Busses

These create a monitoring output for talent, containing a mix of all program elements minus the associated talent’s channel. The mixer supports one or more Mix Minus Busses with independent outputs.

  • Each Mix Minus bus includes a level adjustment after the summing block.
  • They can be configured to include a Compressor, Limiter, and Delay.
  • The Mix minus bus features automatic echo cancellation capabilities.

Aux Busses (Auxiliary Busses)

These provide additional routing paths for effects, monitors, or external processing. The mixer supports one or more independent Aux Busses, which can be Mono, Stereo, or 5.1.

  • Each Aux bus includes a level adjustment after the summing block.
  • They can be configured to include a Compressor, Limiter, and Delay.

Listen Busses (AFL and PFL)

  • AFL (After Fader Listen) Bus: Allows monitoring of audio signal post-fader adjustments. It can be Mono, Stereo, or 5.1. Signals from multiple input channels can be routed simultaneously, and the output includes a dedicated level adjustment.
  • PFL (Pre-Fader Listen) Bus: Allows monitoring of audio signal pre-fader adjustments. It can be Mono, Stereo, or 5.1. Signals from multiple input channels can be routed simultaneously, and the output includes a dedicated level adjustment.

Monitoring and Level Measurement

Monitoring Output Section

 

  • Outputs: Provides a 6-channel output supporting stereo and 5.1, and a 12-channel output for NGA 7.1.4 Multi-Dimensional Monitoring. The 7.1.4 monitoring section includes a specialized matrix for accurately downmixing to 5.1, stereo, and mono.
  • Output Listen: A selector allows the monitoring output to listen to any Program, Subgroup, Mix-Minus, or Aux bus, as well as external signals.
  • Monitoring Logic: The monitor section can automatically switch from the “Output Listen” to the PFL or AFL bus when any input channel or subgroup is routed to them. A “clear function” cancels all active PFLs or AFLs.
  • Controls: Includes Speaker Level Control, Speaker Selector for 3 independent multi-channel outputs/headphone sets (only one active), and individual and simultaneous Mute Functionality for each output channel.
  • Dimming/Talkback: A Speaker Dimming Adjustment dims the monitoring output to a preset level when a dim command is received or when Talkback Functionality is activated.

Input and Output Level Measurement

The Mixer provides components for continuous audio level measurement that feed data via the API for GUI display. These components can be positioned at any point in an input or output processing chain and can measure mono, stereo, or 5.1 audio groupings.

  • Metering: Can be configured to measure peak programme meter (PPM), VU, or instantaneous loudness.
  • Loudness Measurement: Conforms to ITU BS.1770-5 and EBU R 128 standards, providing True Peak, Momentary (400 msec), Short Term (3 sec.), and Integrated readings. The component can be positioned at the output of every Program Bus.
  • Detection: The component can provide an Overload Sensor indication based on a clipping threshold, a Signal Detector indication based on a minimum threshold over a specified duration, and a Silence Detector indication if the signal remains below a threshold for a specified duration. All conditions are communicated via the API.

Control Plane API and Operator GUI

Control Plane API

The API provides comprehensive access to the mixer’s functionality:

  • Read/Write Access: Provides read and write access to all configuration (“set and forget”) and operational parameters.
  • Status Reporting: All metering and signal status information can be read via the API.
  • External Control: Supports both the Mackie Human User Interface MIDI (Mackie HUI protocol) and OSC (Open Sound Control) for control from generic control panels.
  • Presets: Supports independent storage and recall capabilities for Configuration and Operator Parameters (restoring operational settings while leaving configuration unchanged). Selective Parameter Recall is supported for saving and recalling subsets of user settings.
  • Input Channel Presets allow saving and recalling operational settings (gain, dynamics, filtering, EQ) for a specific input channel.
  • The mixer’s conformance to common configuration, control, preset, and health monitoring requirements.

Operator GUI

The Operator GUI is used by operators to control and operate the mixer, offering several key features:

  • Customization and Touchscreen: It is possible to create a custom Operator GUI by combining the controls from individual processing components. GUI components are optimized for touchscreen usage, with select components supporting multi-touch operation and common gesture controls (e.g., sliding two faders, pinching to zoom).
  • Grouping and Linking:
    • The GUI allows operators to configure at least 4 mute groups and assign input channels or subgroups for simultaneous muting.
    • A minimum of 4 VCA groups can be defined, and the GUI includes a fader for each group, which adjusts the overall level of assigned channels while maintaining their relative levels.
    • Fader Linking allows input faders to be linked to other input faders for proportional level adjustment.
  • Workflow Enhancements: Includes capabilities to Colour Code input channels and subgroups, manage Operational Presets (including subset recall), and includes the ability to Copy and Paste a complex set of operational parameters from one channel or processing block to another.
  • Configuration Editor: A WYSIWYG Configuration Editor is available to edit the signal processing configuration, including defining channel inputs/buses/outputs, defining the order/routing of processing functions (“pipelines”), and connecting channel I/O to physical inputs/outputs or other processing resources running on the same platform.
  • Hardware Support: The main operator controls should also be available on a hardware-type control panel, either a dedicated panel or one mapped onto a user-definable control panel.

Sound guy Xmas – ScrewDrivers

You can’t just buy one or two screw drivers for xmas, a whole set is a MUCH better idea.

I’ve had nothing but a fantastic experience with WIHA drivers.

This 33 Piece Drivelock screw driver set is a versatile and lightweight solution to carrying around 33 full size drivers.
33-pc-drive-loc-multi-blade-tray-set

  • Includes Both Soft Cushion Grip & MicroFinish Non-Slip Grip Handles
  • Slotted: 3, 3.5, 4, 4.5, 5.5, 6, 6.5
  • Phillips: #0,#1 & #2
  • Square: 1 & 2
  • TORX: T6,7,8,9,10,15,20, 25,30,40
  • Nut Drivers Inch 3/16, 1/4, 5/16, 11/32, 3/8, 7/16, 1/2
  • Nut Drivers Metric: 5.5, 6, 7, 8, 9, 10,11mm
  • Inch HEX: 3/32, 7/64, 1/8, 9/64, 5/32, 3/16
  • Metric HEX: 2, 2.5, 3, 4, 5, 6mm

If you have never seen the Drive Lock system in action check this out

If the above is too big, a great starting point is the Drive-Loc VI 15 Pc. Combo Set in Pouch They roll up nicely into any gig bag.
drive-loc-vi-15-pc-combo-set-in-pouch-
 Wiha also offer this 51 piece Precision screw driver set that will last your professional a lifetime!
51-piece-precision-screwdrivers-tech-set

 

  • Slotted Tips:  0.8, 1.0, 1.2, 1.5, 1.8, 2.0, 2.5, 3.0, 3.5, 4.0mm
  • Phillips Tips: #000, 00, 0, 1
  • Hex Metric Tips: 1.3, 1.5, 2.0, 2.5, 3.0, 4.0mm
  • Hex Inch Tips: .028, .035, .050, 1/16, 5/64, 3/32, 7/64, 1/8, 9/64, 5/32
  • Torx Tips: T1, 2, 3, 4, 5, 6, 7, 8, 9, & T10
  • Nut Drivers: 1.5, 1.8, 2, 2.5, 3, 3.5, 4, 4.5, 5, 5.5mm

In the same vein as the drive lock drivers, the System 4 screw drivers also offer complete mobility by reducing the weight of screw driver handles.

 

system-4-esd-12-piece-slotted-phillips-torx-tweezers-set

Sound Guy XMAS!!!!! Rigol DS1054Z 4 Channel Oscilloscope –

Xmas is coming!  As we know sound guys are the most difficult people to buy for.  Over the last few year I’ve posted some suggestions, here is day 1!


Day 1 – Rigol DS1054Z 4 Channel Oscilloscope

Screen Shot 2015-11-06 at 8.09.34 PM

This came to my attention from the EEVBLOG https://www.youtube.com/watch?v=ETCOhzU1O5A love this guy.

It’s an awesome price for a very powerful little machine.  Include probes and USB cable.

For $507 Canadian Electro Meter in pickering can hook you up!

 

 

Fixing Logitech trackball performance

I love these Logitech Marble Mouse trackballs, I have one for every computer.  After a few years they slow down physically.  Cleaning the dust off does not help enough, something different needed to be applied.  I’ve been researching lubricants and oils, doing some tests with households; but I think I’ve found the perfect fluid.  Ian Gomes of Union Sound Company made a recommendation a few years back for Penny and Giles faders.  A quick amazon search lead me to Lucas Silicone Shock fluid.  I must report that it is the perfect product for this application.  If you need a drop, I’ll send you a q-tip of it in the mail.

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