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1.1 The Ableton Team Says: Thank You

Live is the result of musicians wanting a better way to create, produce and perform music using a computer. A great deal of effort has been put into making Live easy and fun to use, yet at the same time capable of helping you create music with unlimited depth and sophistication. This effort continues even as you read these lines... in fact, a new, improved Live version might already be available for download! Please check on our website now*http://www.ableton.com/downloads, or choose the Check for Updates command from the Help menu.

We hope you enjoy using Live and that it enhances your creative process.

Your Ableton Team.

1.2 What's New in Live 10?

1.2.1 Device Additions and Improvements

Wavetable (see 24.10) is a new dual-oscillator wavetable synthesizer that's both easy to program and deeply powerful.

Echo (see 22.13) is a new and highly-flexible delay effect with lots of modulation possibilities. It's capable of a wide range of sounds from classic analog-style delays to modern, digital approaches.

Drum Buss (see 22.10) is a new effect that brings analog-style character, body, and glue to drum tracks. Drum Buss combines a range of commonly used drum processing tools into a single device:

Pedal (see 22.28) is a new effect, inspired by three classic guitar distortion pedals. Pedal is capable of a range of distortion sounds from warm overdrive to aggressive distortion to heavy and broken fuzz tones.

Utility's (see 22.36) Gain control can now be adjusted from -infinity dB to +35 dB. The Left/Right Phase buttons now invert the signal at Utility's input rather than at its output. A Bass Mono section allows you to convert frequencies below an adjustable point to mono, with an adjustable crossover control.

EQ Eight (see 22.14) can now boost or cut frequencies down to 10 Hz.

1.2.2 Capture MIDI

Capture MIDI ( see 16.10) introduces a new way of (not) thinking about MIDI recording.

Now, Live is always listening to MIDI input on armed or input-monitored tracks. Press the Capture MIDI button after you've played to create a clip containing your music. Live will automatically try to find the tempo and set appropriate loop boundaries.

With Capture MIDI, you never have to worry about losing your experiments and improvisations because you forgot to start recording. You can always capture what you've just played, after you've played it.

1.2.3 Multi-Clip Editing

Up to eight MIDI clips can be viewed simultaneously in the Detail View.

Within the multi-clip view ( see 10.5), clicking a clip's notes or its loop bar selects this clip for editing.

If multiple clips of different lengths are selected, the editor will show as many loop iterations as are necessary for the clips to realign.

1.2.4 Nested Group Tracks

Group Tracks ( see 15.3) can now be created or moved within other Group Tracks. There is no limit to the number of nested Group Tracks.

1.2.5 Arrangement Editing Improvements

Holding CTRL(PC) / CMD(Mac) while scrolling will zoom to the mouse pointer's position. Holding ALT while scrolling will zoom the amplitude or pitch of the selected tracks.

All Arrangement clips within a time selection can be moved using the left and right arrow keys. Hold ALT(PC) / CMD(Mac) while scrolling to temporarily toggle the grid on or off.

CTRL-Shift(PC) / ALT-Shift(Mac)-dragging within an Arrangement clip will slide the material within the clip without moving the clip itself. HoldALT(PC) / CMD(Mac) while dragging to temporarily toggle the grid on or off.

Holding Shift while dragging the edge of a warped audio clip in the Arrangement will stretch the audio.

Arrangement clips can be moved by dragging the upper half of the clip. Dragging in the lower half will select time.

It's now possible to activate, deactivate, or reverse a selected range of time. If the time selection includes only a portion of a clip or multiple clips, Live will split the clips as necessary, then perform the action on the split clips.

Live now plays back MIDI notes in the Arrangement even if playback is started at a position after the beginning of the notes. This behavior can be toggled off or on via the Chase MIDI Notes command in the Options menu.

Double-clicking in a MIDI track in the Arrangement will create a new clip. The clip will be the length of the Arrangement's grid resolution.

When editing in the Arrangement or Detail View, Follow is now paused rather than disabled. Follow will resume when scrubbing or restarting playback.

CTRL-L(PC) / CMD-L(Mac) will now toggle the Arrangement loop on or off.

Audio or MIDI files can be imported via an "Import..." command in the Create menu. The file will be inserted at the Insert Marker position in the Arrangement View or in the selected clip slot in the Session View.

The Lock Envelopes switch (see 19.5.6) can be MIDI or key mapped.

1.2.6 Automation Improvements

Clip fades are now always available directly on the clip if the track is tall enough.

In the Arrangement, all automation lanes are shown or hidden via the Automation Mode button or the A key. When automation lanes are visible, the content lanes are minimized to the height of the clip headers.

It is possible to Split or Consolidate clips in a time selection within an automation lane.

Introduced fade edge handles in the Arrangement View, which allow users to change the duration of a fade in (via the Fade In Start Handle) and fade out (via the Fade Out End Handle) without affecting the fade peaks. However, a fade edge cannot move beyond the fade peak. To do this, the clip start/end must be used to increase the size of the clip.

Breakpoints now snap to the grid (or to existing breakpoints) when moved horizontally.

Breakpoint values can now be adjusted vertically without moving the breakpoints in time.

Automation segments can be moved horizontally.

It is now possible to create an automation breakpoint by double-clicking anywhere in an envelope editor/automation lane. In the Arrangement, this means that MIDI clips can only be created by double-clicking in the content lane.

Automation values are now displayed when hovering over or dragging breakpoints or automation segments.

The Shift modifier is no longer required for moving automation segments past existing breakpoints.

In Arrangement View, Detail View, Simpler and Sampler, holding CTRL(PC) / CMD(Mac) while scrolling will zoom to the mouse pointer's position.

The Detail View now zooms to show the selected time within an Arrangement clip.

Pressing Z will zoom to the time selection in the Arrangement and Detail Views. Press X to zoom back out.

On U.S. keyboards, it is now possible to zoom by pressing + without holding Shift.

Audio effects can now be dragged to the Master track's Scene Launch area.

Dragging clips from an existing track to the clip/device drop area creates a new track with those clips, and also the devices from the original track.

Dragging frozen clips and tails to the clip/device drop area in the Arrangement View creates a new frozen track (which also contains the devices from the original track).

Holding Shift while scrolling will scroll horizontally on both Mac and Windows (this previously only worked on Macs).

1.2.8 Mixing Improvements

A right-click(PC) / CTRL-click(Mac) command can switch each track's Pan control to a Split Stereo Pan control. In Split Stereo mode, you can adjust the stereo position of the left and right channels independently.

Double-clicking on a knob or slider will reset it to its default value.

Individual Drum Rack pads can now be routed to one of the Drum Rack's internal return chains.

A number of small improvements make mixing easier:

A new Collections section (see "Collections") of the browser allows you to tag browser items with up to seven colors. Each color can also be given a custom name. Searching or browsing within a Collection will then show only items with that tag applied.

Uninstalled Packs or Packs with available updates can be installed directly from the browser.

Group Tracks (including nested Group Tracks) now appear when unfolding Sets in the browser, and can be dragged into the current Set.

1.2.9 Browser Improvements

It is now possible to provide custom names for input and output channels via the Input/Output Config buttons in the Audio Preferences. These names then appear in all Input/Output choosers.

It is now possible to choose between different metronome sounds via the Metronome's pull-down and context menus.

The Metronome can be set to click at various note divisions via its pull-down and context menus. The default Auto setting follows the time signature's denominator. Note divisions that don't fit into one bar of the current time signature will be disabled.

When Enable Only While Recording is enabled in the Metronome's pull-down and context menus, the Metronome will only be audible while recording. The Enable Only While Recording setting is not saved and must be enabled every time Live is started.

The Computer MIDI Keyboard is now off by default, and can be toggled on and off with the M key.

Context menu items now react when the mouse is released, instead of after the initial mouse click. This makes it possible to drag away from a selection to cancel triggering an item, while keeping the context menu open. When hovering over disabled context menu items, they are no longer highlighted.

1.2.10 Setup Improvements

Updated the appearance of clips.

Enabled sample dots on waveforms. When zooming in, it's easier to see where samples are on the Beat Time Ruler.

MIDI notes are now drawn in their clip's color in the Note Editor.

Added an Assign Track Color to Clips menu item to the context menu of track headers. The command applies the track color only to clips in the currently active view (Session or Arrangement).

Incorporated a new font throughout the interface.

Live's dials and sliders now look sharp at any zoom level.

Introduced a set of five new Themes (see 2.2) in place of the previously available Live "Skins".

The clip/device drop area is now always displayed in Session View to enable dragging.

Added Pen Tablet Mode to Live's Look/Feel Preferences. Pen Tablet Mode allows graphic tablets and touch screens to control Live. This replaces the "AbsoluteMouseMode" entry of the options.txt file.

Live now supports HiDPI monitors on Windows 10.

1.2.11 Interface Improvements

The Undo history is no longer cleared when saving a Live Set.

The Export Audio/Video dialog now shows toggles for Encode PCM and Encode MP3. When Encode PCM is on, a lossless audio file is created. WAV, AIFF, and FLAC formats are available for PCM export. When Encode MP3 is on, a CBR 320 kbps MP3 file is created. It is possible to export PCM and MP3 simultaneously. If neither toggle is enabled, the Export button will be disabled.

Saving a Set creates a Backup folder within the Project. This folder contains the 10 most recently saved versions of the Set.

Recorded audio files now include a timestamp in the filename.

Live Sets now load samples faster, which may reduce the Set's loading time.

Large Live Sets can now be closed 5-10 times faster.

1.2.12 File Handling Improvements

Live now supports the Jump List.

When using Live 9 and 10 in parallel, they no longer ask for admin rights if you start them interchangeably.

Owners of both Live 9 and 10 can choose which version of Live to open a Live Set with, by using the right-click(PC) / CTRL-click(Mac) command on that Set and selecting "Open With".

It is now possible to associate .als files with a specific Live version, via "Control Panel" - "Apps" - "Default Apps".

Max for Live is now bundled with the Live installation and is loaded whenever Live is started.

Max for Live devices can now support multiple audio inputs and outputs, and routing is available via the Live API.

Numerous changes for Windows 10:

Compressor's Gain Reduction view and Sidechain controls are now available on Push 2.

EQ Eight's filters and spectrum are now visible on Push 2's display.

Operator's envelopes are now displayed graphically on Push 2's display.

1.2.14 Push Device Visualizations

Melodic Sequencer + 32 Notes is a new note layout on Push 1 (see 28.7) and Push 2 (see 29.7) that uses the bottom 32 pads for notes and the top 32 as a step sequencer. Pressing notes selects them for sequencing. This mode allows for real-time playing and step sequencing at the same time. Holding Note (Push 1) or Layout (Push 2) temporarily shows the loop length controls in the fifth row of pads.

When using the Loop Selector layout in Drum Racks and Simpler's Slicing Mode, holding Note (Push 1) or Layout (Push 2) momentarily toggles the 16 Velocities layout.

When using the 64-pad layout, holding Note (Push 1) or Layout (Push 2) momentarily shows the loop length controls in the top row of pads.

Holding Shift while pressing Note (Push 1) or holding Shift while pressing Layout (Push 2) now locks the alternate layout on. Pressing Note (Push 1) or Layout (Push 2) again returns to the original layout.

Layouts are now remembered per track.

Holding Duplicate and tapping a loop length pad copies the contents of that page and allows you to paste it to another page by tapping the corresponding loop length pad.

Holding Delete and tapping a loop length pad deletes the content of that page.

1.2.15 Push Note Layouts

When using MIDI tracks in Clip Mode (see 29.14), Push 2's display shows the notes within the selected MIDI clip.

MIDI clips can be cropped from Push 2.

1.2.16 Push MIDI Clip Mode

Control Surfaces, including Push 1 and 2, are now available when Live runs as a ReWire device.

The Repeat button's state is now remembered for each track.

Added Split Stereo Pan Mode to Push 1 and 2. When Split Stereo Pan Mode is active while in Global Mix Mode, the current pan value is displayed but the pan dial is inactive. In Track Mix Mode, either the pan dial or stereo pan sliders are shown, depending on the active pan mode.

It is now possible to convert audio to MIDI from Push 2, via the Convert button.

The Collections folder for color labels is now a root level entry in the Push 2 browser.

Nested chains can now be displayed, folded, and unfolded on Push 2.

On Push 2, holding a track's selection button arms the track.

1.2.17 Additional Push Improvements

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Welcome to Live - Ableton Reference Manual Version 10 | Ableton

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Live is the result of musicians wanting a better way to create, produce and perform music using a computer. A great deal of effort has been put into making Live easy and fun to use, yet at the same time capable of helping you create music with unlimited depth and sophistication.

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Live's flexible architecture makes it the perfect choice for scoring to video. You can trim video clips to select parts of them and use Warp Markers to visually align music in the Arrangement View with the video. You can then render ( see 5.2.3) your edited video file along with your audio.

Before diving in, you will want to be familiar with the concepts presented in the Audio Clips, Tempo, and Warping chapter ( see Chapter 9).

If you are interested in syncing Live with external video equipment, you'll also want to read the chapter on synchronization ( see Chapter 30).

(Note: video support is not available in the Intro and Lite Editions.)

21.1 Importing Video

Live can import movies in Apple QuickTime format (.mov) to be used as video clips. Movie files appear in Live's browser and can be imported by dragging them into the Live Set.

Note that Live will only display video for video clips residing in the Arrangement View. Movie files that are loaded into the Session View are treated as audio clips.

21.2 The Appearance of Video in Live

21.2.1 Video Clips in the Arrangement View

A video clip in the Arrangement View looks just like an audio clip, except for the "sprocket holes" in its title bar.

A Video Clip in the Arrangement View.

For the most part, video clips in the Arrangement View are treated just like audio clips. They can be trimmed, for example, by dragging their right or left edges. However, there are some editing commands that, when applied to a video clip, will cause it to be replaced by an audio clip (which by definition has no video component). This replacement only occurs internally - your original movie files are never altered. The commands which will cause this are: Consolidate, Reverse and Crop.

21.2.2 The Video Window

The Video Window in the Arrangement View.

The Video Window is a separate, floating window that always remains above Live's main window. It can be dragged to any location you like, and it will never get covered up by Live. You can toggle its visibility with a command in the View menu. The Video Window can be resized by dragging its bottom right-hand corner. The size and location of this window are not specific to the Set, and will be restored when you open a video again. The video can be shown in full screen (and optionally on a second monitor) by double-clicking in the Video Window. ALT(PC) / ALT(Mac) double-click in the Video Window to restore it to original size of the video.

Movies with Partial Tracks

In the QuickTime file format, the audio and video components do not have to span the entire length of a movie; gaps in playback are allowed. During gaps in video, Live's Video Window will display a black screen; gaps in audio will play silence.

21.2.3 Clip View

Soundtrack composers will want to note the Tempo Leader option in Live's Clip View. When scoring to video, video clips are usually set as tempo leaders, while audio clips are left as tempo followers. These are, therefore, the default warp properties of clips in the Arrangement View. In this scenario, adding Warp Markers to a video clip defines "hit points" that the music will sync to. Note that a video clip's Warp switch needs to be activated in order for the clip to be set as the tempo leader.

Setting a Video Clip as Tempo Leader.

Remember from the Audio Clips, Tempo, and Warping chapter ( see Chapter 9) that, although any number of warped Arrangement clips can have the Tempo Leader option activated, only the bottom-most, currently playing clip is the actual tempo leader.

This also means that it is possible for video clips that are not the current tempo leader to become warped, resulting in warped video output in the Video Window.

Warp Markers

While dragging a Warp Marker belonging to a video clip, you will notice that the Video Window updates to show the corresponding video frame, so that any point in the music can be easily aligned with any point in the video clip.

Since Live displays a movie file's embedded QuickTime markers, they can be used as convenient visual cues when setting Warp Markers.

21.3 Matching Sound to Video

In Live, it takes just a few steps to get started with video. Let's look at a common scenario - matching a piece of music to edits or hit points in a video:

Make sure that Live's Arrangement View is visible. If you're using Live on a single monitor, your computer keyboard's Tab key will toggle between the Session View and Arrangement View.

Drag a QuickTime movie from Live's browser and drop it into an audio track in the Arrangement View. The Video Window will appear to display the video component of the movie file. (Remember that you can move this window to any convenient location on the screen.)

Now that the video clip is loaded, drag an audio clip into the Arrangement View's drop area. A new track will automatically be created for it. Unfold both tracks so you can see their contents by clicking the buttons to the left of their names.

Double-click on the video clip's title bar to view it in the Clip View. In the Sample box, make sure that the Warp button is enabled. Warped clips in the Arrangement View can be set as tempo leader or follower. We want the Leader/Follower switch set to Leader, which will force the rest of the clips in the Live Set to adapt to the video clip's tempo (i.e., its normal playback rate).

Now add Warp Markers to the video clip, and adjust them to your liking. The locations of the Warp Markers define the synchronizing points between our music and our video. Notice how the video clip's waveform in the Arrangement View updates to reflect your changes as you make them.

If desired, enable the Arrangement Loop (see 6.5) to focus on a specific section of the composition.

When you have finished, choose the Export Audio/Video command from Live's File menu. All of your audio will be mixed down and saved as a single audio file. You can also export your video file using this command.

21.4 Video Trimming Tricks

Commonly, composers receive movie files with a few seconds of blank space before the "real" beginning of the action. This pre-roll ("two-beep") serves as a sync reference for the mixing engineer, who expects that the composer's audio files will also include the same pre-roll. While working on music, however, the pre-roll is in the composer's way: It would be more natural for the movie action to start at song time 1.1.1 and SMPTE time 00:00:00:00. This can be accommodated by trimming video clips, as follows.

First, we drop a movie file at the start of the Arrangement (1.1.1).

A Video Clip at the Start of the Arrangement.

Next, we double-click on the video clip's title bar to display its contents in the Clip View. There, we drag the Start Marker to the right so the video clip starts at the beginning of the action.

Dragging the Start Marker behind the Pre-Roll.

Now, both the action and the music to be composed start at 1.1.1 / 00.00.00.00. Once the music is done and ready to be rendered to disk, we need to bring back the pre-roll:

In the Arrangement View, we select all materials (Edit menu/Select All), then drag the entire composition a few seconds to the right:

The Video Clip and the Final Clip of Music.

Now, we click on the video clip's title bar (to deselect everything else), then drag the video clip's left edge to the left as far as possible to reveal the pre-roll again.

The Video Clip with Pre-Roll Restored.

The Export Audio/Video command, by default, creates sample files as long as the Arrangement selection; as the video clip is still selected, the exported sample file will have the exact same duration as the original movie file, including the pre-roll.

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Working with Video - Ableton Reference Manual Version 10 | Ableton

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Live's flexible architecture makes it the perfect choice for scoring to video. You can trim video clips to select parts of them and use Warp Markers to visually align music in the Arrangement View with the video. You can then render ( see 5.2.3) your edited video file along with your audio.

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Composer Magazine is a platform for inspiration showcasing emerging artists and interviews with top composers writing the latest Hollywood blockbuster film scores.

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The same pitfalls present themselves time and again in project-studio recording. Learn to avoid them and your tracking sessions will be transformed!

As a long-time contributor to SOS's Mix Rescue column, I've tried to demonstrate that, with careful editing and processing, it's often possible to snatch victory from the jaws of defeat when faced with poor recordings. But long, tedious salvage jobs at the mix would be unnecessary if the recordings were better in the first place. A few simple changes while recording could have saved me many hours of post-production purgatory. I wouldn't expect a well-recorded band production to take more than a day or so to mix to a commercial level, but most project-studio multitracks I receive need three times that amount of work, and some Mix Rescue projects can take a solid week to disentangle. What's most galling is that the same basic tracking errors are to blame time and time again, so with this in mind, I'd like to help you cut hours (or perhaps days) off your mixing schedule. I'll list the 10 most common recording traps I hear project-studio producers fall into, and explain some no-nonsense ways to avoid them. To help you understand things better, I've put together a media page on the SOS web site ( http://sosm.ag/feb16media) which features dozens of audio demonstrations and provides links to a variety of helpful online resources. Also, in case any of the technical terms I use leaves you scratching your head, I've included links to the capacious SOS technical glossary and numerous supporting articles from our free online archive. Now, let's consider those mistakes...
Quick Tips...
Work on the source: Listen acoustically to the instrument/ensemble you're about to record and ask yourself if it already sounds the way you want it to. If not, always try to address this at source - not only before you hit Record, but before you even get any mics involved. As producer Trina Shoemaker puts it: "It's not about getting a great sound. It's about capturing a great sound that's already there." Clearly, you may never reach your perfect timbre that way, especially when working under typical project-studio restrictions, but the process of striving for the unattainable ideal makes it much more likely that you'll get a respectable tone through the mics.

Consider the context: Soloing a source can help you home in on technical problems, but a beautiful sound in isolation won't necessarily fit within the context of a full arrangement. Don't make hard-and-fast decisions about any instrument's tone while your solo buttons are lit.

Ask the musician: Whenever your mic setup doesn't immediately strike gold (and certainly before you start reaching for EQ), quickly record a snippet, play it back for the musician, and ask them to help you improve the sound. They'll almost certainly come up with half a dozen great suggestions straight away: an electric guitarist might flick a switch or use a different pick; a keyboard player might rearrange his chord voicings or tweak some arcane synth parameter; or a singer might increase their twang or breathiness. No engineer can know how every instrument works, so why not just ask the expert who's playing it? Besides, encouraging performers to critique their own recorded sound is no bad thing in general, because they'll often respond instinctively to it and adjust their playing to fit the arrangement - in effect doing some of the mixing work for you!

Help the vocalist: Encourage singers to sip room-temperature water between vocal takes. The time of day you record can make a huge difference to the sound of any voice, and the tone will also change as the singer warms up, hits their peak, and then gradually fatigues. With a little planning, you can turn these things to your advantage.

Work the room: Don't forget to exploit the acoustic potential of your surroundings. Is there another spot in the recording room that might sound better, or another room entirely that you could decamp to? What if you removed the comfy chairs and rolled up the carpet to enhance an acoustic-guitar sound, or hung up bed quilts to tackle some harsh spill between two instruments? These things are free, for Heaven's sake, and often alter the sonics more profoundly than your mics ever could, so don't squander them!

Check tuning often: Check the tuning before every take. (Pretty please?!) Honestly, it's only a minute's work and it can save countless hours of soul-crushing corrective editing later on.

One big problem is that few templates are accompanied with unprocessed audio examples, so there's no way of knowing whether the suggested mic technique will actually suit your music. Furthermore, some templates seem also to have been inherited from practices in large commercial studios (and indeed from the heyday of analogue tape recording), and frequently crash and burn when you're working in more domestic acoustics, typically in a smaller space and with a limited amount of affordable gear. I can't tell you, too, how often I see on-stage miking setups being transferred unthinkingly into the studio, even though these tend to heavily prioritise feedback minimisation over sound quality!

Irrespective of the suitability of any miking template, a basic truth of mic technique is that it's not fundamentally about where you put the mics in the first instance. What really matters is how you adjust those initial guesses in response to what you hear through your monitors.
Quick Tips...
Always tweak your mics: I'd estimate that 90 percent of the initial mic choices and placements on typical project-studio sessions would benefit from some adjustment, so changing and/or moving each mic before recording should be the rule, not the exception - whether you start from a miking template or not. Avoiding EQ while recording can help with discipline here. I also prefer to remove all chairs from in front of the recording workstation while I'm setting up for recording, simply because it's a lot harder to motivate myself to keep moving mics around if I'm having to get up out of a comfy seat every time.

Make comparisons: Always make a point of listening acoustically to any sound you're recording. This is something that celebrated engineers stress repeatedly. "Don't just sit down there in the control room," says Bruce Swedien, for instance. "Go see what it sounds like in the studio." One of the best weapons against template-driven complacency is the discipline of comparing your monitored signal with acoustic reality. It can be a bit depressing at times, but there's nothing else that'll hone your recording chops as effectively.

Fix it before the mix: One of the biggest danger signs when people are using templates is the excuse: "well, it does still need to be mixed". Yes, the purpose of mixing is to make a recording sound its best, but if your rough tracking balance isn't getting the musicians excited, then you unquestionably need to work harder with the mics. It's a massive waste of time trying to 'fix things in the mix', because you have many more options to manipulate timbres while recording. As Trevor Horn once put it: "The mix is the worst time to do anything."

Use your ears: But if miking templates aren't much use, how else can you make sensible initial miking choices for your sessions? Well, the best way is simply to use your ears! If you're wondering what kind of mic to use, for instance, it's child's play to find microphone shootouts on the Internet - for example, here at SOS we've done our own carefully controlled comparisons of ribbon mics ( http://sosm.ag/ribbonmics) and vocal mics ( http://sosm.ag/voxmics). Pure microphone placement demonstrations (ie. multiple instances of the same mic on a single performance) are rather less common, so to put that right, I've recently begun building up a Library of Microphone Positions ( www.cambridge-mt.com/rs-lmp.htm. This already features demonstration files for stalwarts such as electric and acoustic guitars, grand and upright pianos, kick, snare, cymbals, upright bass and saxophone.

The demise of so many large professional facilities means that few young engineers now benefit from the traditional 'tea-boy to tape-op' studio apprenticeship, and hence seldom have first-hand opportunities to watch seasoned veterans at work. Although there's a lot you can learn nowadays without all that sharpening of pencils and emptying of ashtrays, one thing that's difficult to convey is just how critical professionals are in the studio, particularly when it comes to the time and effort they take to develop and edit performances.

Top-quality lead vocals are probably the clearest indicator of a professional-grade production in this respect, but I'd say nine out of 10 project-studio mixes fail to deliver that quality level, simply because there's been skimping at the tracking and editing stages.
Quick Tips...
Listen, appraise & act: Whenever you capture anything, make a point of listening back to the recording with a critical frame of mind. For example, are there any small errors that could do with patching over? Is the second verse performed as musically as the first? Are there any moments where the groove suddenly seems to hurry or drag? Get to know the keyboard shortcut for adding marker points in your DAW project, and use it to highlight any questionable moments as you listen through to the take, so you don't forget anything. Acquaint yourself with your DAW's manual punch-in facility as well (and its associated cue-mix switching functions), because it's usually quicker to patch over blemishes on the fly than by doing longer takes and then editing separately.

Compare takes: For a lead vocal in any chart style I'd recommend recording a minimum of four takes, and then comping a master take from the best bits of each. Many mainstream commercial productions go way beyond that, though, assembling more than a dozen takes and devoting hours to the comping process. Whatever the degree of depth you go into, though, here's one time-saving suggestion: don't just keep recording new takes without listening back to the older ones. Early takes will often be outshone by later ones, in which case they can be recorded over to reduce your editing workload. In addition, I find that comparing takes during the tracking session itself introduces an element of self-criticism and competitiveness that frequently spurs vocalists to up their game.

Resist making do: Be wary of any situation in which you find yourself uttering the words "that'll do" or "we'll fix that later". Another danger sign is if you feel an insatiable urge to add extra overdubs, because that often indicates that the existing parts aren't yet compelling enough in their own right. Broadly speaking, the better your performances, the easier it is to leave your arrangement sparse. In a similar vein, try to resist sweetening your tracking rough mix with masses of reverb or delay effects, since these make second-rate moments in a performance tougher to spot.
Quick Tips...
Beware bright mics: Be careful of reusing mic setups developed during the heyday of analogue tape, because high frequencies were often pre-emphasised at the tracking stage, using super-bright capacitor mics as a way of compensating for treble losses in the recording medium. "If you're going to record on a digital medium," comments producer Mick Glossop, for instance, "you can't expect to use the same mics you've used for analogue." The recent renaissance of ribbon mics in pro circles is a testament to this, because ribbons deliver a much smoother sound at source, but project-studio owners still seem reluctant to follow suit, despite the recent price crash in the ribbon-mic market. Perhaps it's because they're wary of the figure-of-eight polar patterns, which most ribbons exhibit. Whatever the reason, I suspect this is why drum overheads, acoustic guitar, strings, and horns so often sound painfully tinny in home-brew mixes.

Try different mics before EQ'ing: While recording, think like a mix engineer. But rather than using EQ to shape the sound, use your choice of mic, with the aim of contrasting different sounds so that they don't conflict and obscure each other at mixdown. This is where dynamic designs really come into their own for me, because you'll typically get a much wider tonal palette from spending a given mic budget on dynamic mics than on condensers. Granted, most recordists usually have a Shure SM57 or SM58 and an AKG D112 or some such lying around, but there are plenty of other stellar performers in the £250-£500 $350-700 price range - mics such as the Beyerdynamic M88, Electro-Voice RE20 and RE320, Heil Sound PR30 and PR40, Sennheiser MD421 and MD441, and Shure SM7B, to name just a handful.

Think of the transients: Another thing tape-based recorders have trouble retaining is transient definition, and it wasn't uncommon to see hand percussion recorded close up with a small-diaphragm capacitor mic to compensate for this. On digital systems, such a tactic can be disastrous because all you end up hearing is the transients, rather than any of the body of the instrument's timbre. Dynamic or ribbon mics are more understated in this respect, so they'll usually give you a fatter, more mixable sound right from the outset.

Pick mics for each vocalist: Amateur engineers are particularly prone to choosing capacitor mics automatically when recording lead vocals, which is a shame, since harsh, over-bright vocal sounds are a problem I commonly encounter in Mix Rescues, and the problem could easily have been solved in most cases by using a dynamic mic. Bear in mind that many iconic studio vocal performances were captured with dynamic mics. The Shure SM57 and SM58 have a long history on record with powerful rock vocalists, but that's only the tip of the iceberg. Beyerdynamic's M88 was Phil Collins' mic of choice for years, for instance, and Michael Jackson used a Shure SM7 for several cuts on Thriller. Electro-Voice's RE20 has been another perennial vocal favourite, capturing female vocals on Fleetwood Mac's Rumours and Stevie Wonder's lead on 'I Wish', for example.
Quick Tips...
Check for cue-mix problems: When a performer is struggling with tuning while overdubbing, it may be the foldback mix at fault, so ask to listen to their headphones. The most common culprits are the backing-track being too quiet, so the performer can't clearly hear pitch cues from other instruments, or the performer's own signal being too quiet, so they can't judge their own pitch reliably. With vocalists, another big factor is monitoring latency, because even the slightest latency delay incurs comb-filtering between the monitoring signal and sound vibrations carried directly to the singer's ear via bone conduction. The simplest workaround is to encourage singers to monitor themselves acoustically by slipping one headphone earcup off their ear, but it's usually better in the long run to set up zero-latency (ie. analogue) monitoring, which eliminates the monitoring delay completely.

Look for signs of unease: A tentative performance can signal that the performer is mixed uncomfortably loud in their own cans, whereas a musician who can't hear themselves well enough will often end up over-projecting, delivering a strained and spindly tone and succumbing more quickly to fatigue. Performers are frequently reticent about asking for headphone mix changes, so stay alert for the sonic warning signs.

Give each performer the balance they need: When recording ensembles, try to resist the urge to give everyone headphones from the get-go. Instead, consider who really needs them (and indeed what each person actually needs to hear) and work from there. If your band wants to use a click track, for example, maybe just give that to the drummer and have everyone else follow him/her, as they would on stage. Or if you have to isolate a band's singer in a separate room, feed vocal-only foldback to the rest of the band, rather than giving them each their own full-band mix - each player should be able to hear their own natural band balance by using just one headphone earcup, assuming you've not set the instruments up too far apart.

Consider losing the cans: Don't neglect the possibility of loudspeaker foldback when overdubbing, especially with fledgling studio vocalists for whom headphone monitoring can be a bit unsettling. Normally I wouldn't put any vocal foldback in the monitoring feed in this situation, but if the singer complains that they can't hear themselves well enough, try reflecting some of their own acoustic sound back at them in the first instance using a reflector on one side of the microphone. (A clipboard on a music stand does the job rather nicely, as it happens.) A similar strategy can help singers hear themselves better in an ensemble-recording situation, too.
Quick Tips...
Don't mic too wide or too close: To avoid an over-wide image, don't space the mics in an A-B mic pair more than about 60cm apart, and keep the mutual angle between crossed cardioid mics below 130 degrees. Whichever stereo mic array you choose, be wary of positioning it too close, because the recording angles of many common setups aren't actually very wide in practice. For example, the useful pickup zone of ORTF is roughly 110 degrees, in my experience, while a 60cm-wide A-B configuration will only suit a soundstage of around 60 degrees. The Blumlein array's crossed figure-eights often catch people out here, especially when miking large instruments such as drum kits and pianos, because their useful pickup angle is only 70 degrees or so, and any sound source located outside the frontal 90-degree arc will suffer tremendous mono-compatibility problems.

Check references when using headphones: It's almost impossible to judge stereo width instinctively over headphones, because of the way they stretch the recorded panorama across a 180-degree angle. To improve your decision-making, keep a couple of suitable commercial recordings on hand for comparison and use a stereo vectorscope as a visual confidence check. Plus, don't forget to switch the mix to mono from time to time to check for unacceptable timbral side-effects.

Move the instruments: Assuming you're already capturing a suitable image width, the best fix for ensemble imbalances is to adjust the positions of the instruments in relation to the mics. If musicans are set up in a straight line across the sound stage, that'll cause centre-stage sources to appear about 3dB stronger than those at ±30 degrees, so try to avoid that kind of setup if at all possible - where it's unavoidable, you may need to use additional 'outrigger' mics to support the instruments at the edge of the group.

Understand polar patterns: Using cardioids crossed at right angles will overemphasise the level of central sources, so I'd widen the angle to at least 110 degrees for ensemble recording, which gives a recording angle of about 160 degrees. If you choose a Mid-Sides setup with a cardioid Mid mic, it'll tend to give its most natural balance across the soundstage if you treat its pickup angle as being roughly 130 degrees, whereas with a figure-of-eight Mid mic, that angle shrinks to around 90 degrees.

One of the most fundamental rules of recording is beautifully summarised by engineer and author Mike Stavrou: "Don't ever forget - [the musician's] job is more difficult than ours... If our recording setup ruins his ability to perform, we're not helping." Although this seems like common sense to me (and hopefully to you too), I hear way too many horror stories from musicians about engineers putting the cart before the horse, unilaterally imposing some working method which destroys everyone's ability to perform well. There are few better ways to throw a wet blanket over a session than antagonising the talent in that manner and, on a commercial level, I scarcely need add that it's unlikely to generate any repeat custom for the engineer!
Quick Tips...
If musicians prefer to play together, find a way: As a rule, band musicians hate building up productions one overdub at a time. Tracking proceeds at a snail's pace, and often involves a lot of boring waiting around while other musicians record; it's much more difficult to play in time and generate natural musical interactions between different parts when you're not performing in a group; and mixdown is usually a painful process, because earlier tracks need more processing to fit with later additions, and nothing has a natural acoustic connection. Yes, it's more challenging for the engineer to record people playing together, especially in a small studio, but that's really no excuse. Where there's a will, there's a way.

Attend rehearsals: Before you record any young band, visit them in rehearsal. Make a note of where everyone has set themselves up and roughly what each player is seeing and hearing from where they're standing. As you plan your studio setup, think carefully before deciding to deviate from this layout. However, if an otherwise promising session layout jeopardises sight lines, you can sometimes work around that using mirrors. I like to use universal lorry wing mirrors, because they're fairly cheap (£10-£15 $25-30 if you shop around), usefully large, and have rear mounting brackets which fix easily to spare mic stands.

Beware isolating performers: When recording ensembles, try not to distance or isolate players from each other any more than you absolutely have to, because it's difficult for musicians to perform well if they feel physically dislocated from the action. Whatever initial setup you come up with, don't commit to it until you've asked the musicians how it feels for them. In the grand scheme of things, there's a lot you can do to manage spill during recording and mixdown, but you're never going to remedy a shaky performance. "I don't care whether it's classical or rock or country," says producer Eddie Kramer, for instance, "you've got to capture that performance, and the hell with the bloody leakage!"

Make communication easy: While overdubbing, it's a lot easier to communicate with the talent if you're both in the same room. With amplified electric instruments, a long cable from the amp to the instrument (or, even better, from the amp to the speaker cabinet) makes this fairly straightforward, and also allows the player to monitor over any control-room speakers you may have. With acoustic instruments and vocals, a dry sound can usually be captured fine in any control room (even an improvised one) with the help of a duvet or two, but don't disregard the idea of setting up your recording system (or just the keyboard, mouse, and monitor of a computer system via a KVM extension unit) in the live room either, if a more open sound is desirable.
Quick Tips...
Point the instruments elsewhere: Some sound sources, such as electric guitar cabinets, woodwind and brass instruments, are inherently quite directional, and you can use this to your advantage while setting up. For instance, just pointing a trumpeter 30 degrees to one side can make a massive difference to the amount of trumpet spill picked up on another mic, without substantially cramping the musician's style.

Get the balance right: The big trick to getting decent separation between mics is to get the instruments themselves balancing sensibly in the room. So if there's too much guitar spill on the drums, try turning the guitar amps down in the first instance, or alternatively feed the drums through a PA system in the room. Or if the snare drum's too loud on your piano mics, maybe ask the performer to try different sticks, or even brushes. Where acoustic guitar or upright bass are struggling to compete with other instruments acoustically (such that their microphones capture more spill), you can hedge your bets by capturing a DI feed alongside the mic signal, because this allows you to add more 'dry' level at mixdown if necessary. This is a trick Frank Filipetti used on his Grammy-winning recording for James Taylor's Hourglass, so there's no shame in it - although it will sound most natural if you use as little of the DI signal as you can get away with.

Contain reflected spill: Reflected spill is typically more problematic at mixdown than spill coming direct from an instrument, so if you put up acoustic absorbers to reduce spill levels, it's usually better to put those up around the performers, rather than between them.

Check polarity as you go: As you build up your own monitor mix during tracking, be sure to check the polarity setting for each mic/DI signal you add, because the sonic difference this makes can be enormous. There's no 'right' setting, though - just choose the one that sounds best.

Build from the spill: Try to bear in mind how each instrument sounds in your tracking mix before you've added its dedicated mic (ie. when you're only hearing that instrument as spill from other mics), and adjust your miking choices accordingly. If you've already got bright hi-hat spill coming down your snare mic, say, then maybe a meatier-sounding dynamic mic on the hi-hat might make sense. In this context, it makes a lot of sense to start building your rough mix with those mics which capture the most spill, as these will likely have the greatest influence on your mic technique across the session as a whole.
Quick Tips...
Work closer to omni mics: The normal home-studio advice for getting a comparatively dry sound out of a cardioid mic is to hang a thick duvet around the back and sides of the performer to intercept room reflections before they reach the sensitive side of the mic. By extension, you need another quilt to defend the back of a figure-eight mic, and a full enclosure for an omni. (My favourite fixing solution here is to secure little key rings to the blankets, and then hang them on cuphooks screwed into the ceiling.) Remember, though, that omni microphones exhibit very little proximity effect and are fairly insensitive to air blasts (such as plosive vocal consonants), so you can also work closer with them to get a drier pickup.

The pattern can affect tonality: Switching a large-diaphragm capacitor mic's polar pattern doesn't just change its directionality. It can also have a dramatic effect on its tone, especially with budget-friendly mics - and not just on account of the differing amounts of proximity-effect bass boost (which is greatest for figure-of-eight mics and least for omnis). For example, I was recently helping an SOS reader close-mic an acoustic guitar using their Rode NT2A, and the cardioid polar pattern was, frankly, rather brash-sounding under those circumstances. By contrast, the omni pattern might as well have been a different mic - it was that much smoother-sounding!

Pick a spill-friendly pattern: Wherever off-axis sound, such as room ambience or spill, features heavily in your recording, be wary of using large-diaphragm capacitor mics in their cardioid mode. This will often be the least good-sounding option, as sounds arriving from the back and sides of that pickup pattern typically sound rather unpleasant, whereas omni and figure-of-eight patterns generally provide a much more benign tone from all angles.

If a performer moves, try an omni: An omni mic's broader pickup angle, more even off-axis frequency response and negligible proximity effect make it a good choice for maintaining an even tone when instrumentalists or vocalists move around a lot while performing - this is something top-name producers such as Phil Ramone, Al Schmitt, Bruce Swedien, and Michael Tarsia have all mentioned in interview. For similar reasons, an omni is also great when you're recording a comparatively large instrument (such as a piano or acoustic guitar) from very close up.

Choose mics to reject sounds: If you want to mitigate some unwanted sound while recording, whether it's fan noise from your computer, slapback echo from a nearby wall, or spill from a foldback loudspeaker or another instrument, the rejection plane of a figure-of-eight microphone easily trumps the rear null of a cardiod in terms of performance. A pair of figure-of-eight mics can provide astonishing separation when recording singing acoustic guitarists or duetting lead vocalists, for instance, while a 'curtain' of figure-of-eights can be handy for picking up a choir in large-scale orchestral works, because the mics can be angled to reject the brass section blaring away directly in front of the rear risers - a trick Andy Payne mentioned using for his BBC Proms concert recordings in SOS November 2014.

There continues to be a persistent belief amongst home-studio users that the purpose of tracking is to maximise the number of sonic options available when mixing. As reasonable as this may sound in theory, it's often extraordinarily inefficient in practice. The first problem is that 'keeping options open' is usually code for 'shirking decisions about how the production's meant to sound', which makes it well-nigh impossible to choose a suitable recording technique. How can you judge when your mic position sounds right if you've no clear sonic goal in mind? The second problem is that every option you keep open spawns a multitude of others. Let's say you've used two mics and a DI on your bass cab without committing to a balance for them. How many mics will you need on each guitar overdub to make sure its sound can be fitted in with the bass at mixdown? Three? Four? And then what about the piano, vocals, egg shaker...? In short, sonic indecisiveness is a recipe for bloated multitrack projects that you end up loathing once you've finally trudged your way through the inevitable sweat-soaked mixdown nightmare.
Quick Tips...
Bounce things down: Here's a good method for encouraging a more decisive attitude when you're recording any instrument with multiple mics, or with a combination of mic and DI box: make a habit of bouncing down a representative balance of those signals straight away, to a single mono or stereo track. An additional benefit of this approach is that it keeps the total number of tracks down, which speeds up navigating around the session. It's also a good idea to bounce the output of guitar-amp simulators and live-running virtual instruments as audio at the earliest opportunity, which will help free up CPU power for running your audio driver at low latencies without glitches.

In both cases, you can always keep the original source tracks archived somewhere, on the off-chance you really do want to revisit them. Just make sure you don't leave them cluttering up the main recording project.

Edit before overdubbing: Don't overdub new parts in an arrangement until all necessary editing/comping of the existing parts has been completed. This isn't just about committing to decisions, as it also invariably leaves you with better recordings. If you don't address the timing issues of your drum and bass parts, say, before adding guitars and keyboard, you'll end up having to edit the timing of the guitars and keyboards too, regardless of how good the players are. Get the drums and bass groove right from the outset, and there's a much higher probability that the guitars and keyboards won't need editing at all - and you'll have saved yourself a few hours' work. By the same token, don't put a click track in a musician's mix while overdubbing if there are already enough recorded parts to provide an adequate timing reference. The end listener won't hear the click, remember, so it's more important that the musical parts agree rhythmically amongst themselves than that everything lines up exactly with your DAW's notional bars/beats grid.

Don't track the vocals last: Consider recording your lead vocals as soon as possible in the recording session, rather than leaving them to the end of the tracking process. You'd be amazed how much easier this often makes it both to select appropriate sounds for the remaining backing parts, and also to mix the final arrangement once it's complete.

Mix as you go: Another way to maintain a decisive frame of mind (and indeed to improve your mixing instincts) is to keep doing new five-minute rough mixes of your in-progress project every time you record a new part. It might seem odd reinventing the wheel like this, but the advantage is that it encourages you to repeatedly question the role of each existing part in the mix, so you're much more likely to eliminate dead weight from the arrangement and develop a strong vision for the final product. Eric 'Mixerman' Sarafin makes this point strongly in his book Zen & The Art Of Producing: "If, upon completing your last overdub, you have no earthly idea how you want your tracks balanced, then you have been completely ignoring how the track is supposed to make you move and feel... Therefore, you should be toying with your balances each and every time you touch the track."

Mike Senior has worked with various artists including the Charlatans, Reef, Therapy?, Nigel Kennedy, and Wet Wet Wet. He spent several years in the editorial department of Sound On Sound before leaving to take up work as a freelance engineer. He's also written a couple of audio-production books that have received rave reviews ( Mixing Secrets For The Small Studio and, more recently, Recording Secrets For The Small Studio) and put lots of useful additional resources up on his Cambridge Music Technology web site.
www.cambridge-mt.com/RecordingSecrets.htm

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The same pitfalls present themselves time and again in project-studio recording. Learn to avoid them and your tracking sessions will be transformed! As a long-time contributor to SOS's Mix Rescue column, I've tried to demonstrate that, with careful editing and processing, it's often possible to snatch victory from the jaws of defeat when faced with poor recordings.

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Preset synth patches are either a boon or a bane, depending on one's programming prowess. For many 'serious' synthesists, they represent something of a cop-out - easy selections of spoon-fed solutions to impress the inebriated denizens of the local Holiday Inn lounge.
However, the vast majority of keyboard players depend on presets to compose and record their arrangements.

It's not a coincidence that the first synth ever to enjoy truly blockbuster sales was packed with excellent presets. Indeed, Yamaha's fabled DX7 was also notoriously difficult to program, and virtually single-handedly kicked off the third-party patch industry.

It's often the case that an instrument is defined by a handful of presets - Korg's Wavestation being a good example. Chocked to the brim with complex, motion-filled, rhythmic soundscapes, it's a safe bet that few people bought a Wavestation for its bass and Clav sounds.

Some presets have been so heavily used (and abused) that they have become instantly recognisable, shouting above the din of the popular music that they helped to define. Let's take a look, then, at 10 of the most famous preset patches, the synths on which they were supplied, and the songs that burned them into our cultural consciousness.

1. Sync (from the Sequential Circuits Prophet-5)

Before Sequential Circuits unleashed the Prophet-5 in 1977, synthesizer presets were a rarity - most often relegated to non-programmable instruments like ARP's Pro Soloist. Sequential's slick combination of patch storage and polyphony ensured the P5's popularity, with a then-impressive 8000 units being sold, many to big name performers.
Released in 1975, the original Polymoog was a 71-note synthesizer that offered organ-style divide-down polyphony and a miserly eight presets. Troubled from the start, this cantankerous instrument nevertheless found favour with a number of high-profile users, Rick Wakeman among them.
One such player was Greg Hawkes of The Cars, who promptly exploited one of the Prophet's wicked oscillator sync presets for the memorable riff that propels the band's Let's Go, the lead single from their classic Candy-O LP. This ripping, nasal lead patch would - like many of the Prophet's presets - become wildly overused in the years following the instrument's release.

2. Vox Humana (from the Moog Polymoog Keyboard)

It was, however, the second, pared-down version - the Polymoog Keyboard - that would burn its sonic signature into synthesizer history, thanks to one its additional presets - the fabled Vox Humana patch.

This sound was much-loved by Gary Numan, figuring prominently on The Pleasure Principle. It is, in fact, the very first sound on that album. Ideal for mournful leads and low, scrubbing string pads alike, it's easy to see the appeal.

3. PPG Choir (from the PPG Wave 2.2/2.3)

When Wolfgang Palm unleashed his PPG Wave synthesizers back in 1981, few had heard their like before. A devilishly clever combination of digital wavetable oscillators and analogue filters, only the moneyed elite could afford them.

Consequently, though relatively few of them were sold, those that were seemed destined for the charts. Many of the instruments' presets would become identified with the New Wave of the '80s, the famous PPG Choir being but one example.

Tangerine Dream used the haunting metallic vocal patch on their epochal Exit LP, and Depeche Mode would drop it into See You, from their second LP, A Broken Frame.

4. E. PIANO 1 (from the Yamaha DX7)

Few synthesizers have ever had such a lasting impact as Yamaha's DX7. From its stellar playability to its sterile, knob-free front panel, it was nothing like the analogue instruments that dominated the synthesizer market in 1983.

It certainly didn't sound like anything else. With its crisp, spiky percussion, crystalline bells and clinical basses, it was the perfect antidote to a decade of analogue waveforms.

Unfortunately, the instrument that ushered in digital synthesis would bring with it a nearly impenetrable architecture consisting of operators, algorithms and unusual envelopes, all accessed through tedious menus and a diminutive display. Good thing it was loaded with presets, then. Indeed, most DX7 players never bothered making their own sounds, and a third-party patch industry grew up around the instrument.

That didn't stop far too many balladeers from employing the now-famous DX Rhodes emulation, a sound that would become an obligatory inclusion in every keyboard workstation for decades to come.

5. What The (Hoover) (from the Roland Alpha Juno)

Upon its release in 1985, Roland's Alpha Juno was a harbinger of things to come. Retaining the Juno name that had previously been associated with inexpensive, slider-laden polysynths, the spartan front panel of the Alpha Juno lacked the immediacy of its predecessors. Nevertheless, at $895, it was more affordable than the Juno 106 that came before it.

Upon its release, it found favour with many a cash-strapped musician, but it would be future secondhand buyers who would hone in on a single preset called What The - a buzzing sonorous blast that would become widely known as the 'Hoover'.

6. Digital Native Dance (from the Roland D-50)

Wildly successful upon its release in 1987, Roland's D-50 offered a new take on subtractive synthesis that combined sampled attack transients, synthesized waveforms and a smattering of looped samples that could be mixed, filtered and processed by built-in EQ, chorus and reverb effects. It was a brilliant recipe, gobbled up by a staggering 200,000 musicians worldwide.

Subsequently, a number of D-50 presets wound up on the airwaves, though none so instantly recognisable as Digital Native Dance, a skittering atmospheric patch that started life as a novelty whipped up by one of Roland's PCM engineers. Lead patch programmers Eric Persing and Adrian Scott were duly impressed, using the loop to create what would become the D-50's signature sound, which would be used by everyone from OMD to Miles Davis.

7. LatelyBass (Yamaha TX81Z)

The 4-operator TX81Z was yet another synthesizer that was largely unappreciated in its day only to become popular among dance music producers years after its release.

Much of its newfound notoriety rests squarely on this biting bass sound. Unashamedly digital, the LatelyBass patch would wind up in countless Italo disco tracks as well as a fair few pop songs. Producer Eliot (Spice Girls) Kennedy was reputed to keep a TX81Z around solely for this sound, while Babyface's dual, detuned TX81Zs are said to be permanently dialed in to the patch. And, of course, there's Madonna's Vogue.

8. Universe (from the Korg M1)

Released in 1988, Korg's M1 forever changed the synthesizer industry. Designed as an all-in-one workstation, the M1 offered multisampled instruments of unprecedented realism and matched them up with a sophisticated multitrack sequencer and a comprehensive selection of digital effects.

Never before had so much production power been crammed into one instrument. Relatively convincing pianos, organs, basses and guitars, full drum kits with velocity sensitivity, and loads of evocative synthetic timbres made the M1 the synthesizer to own.

It sported plenty of now-cliched presets, including Universe, one of the finest choir patches anyone had heard at the time. Even Queen - who once proudly proclaimed their records free of synthesizers - couldn't resist the call.

9. Ski Jam/The Wave Song (from the Korg Wavestation)

If any ROMpler had the power to transcend the limitations of sample-playback synthesis, it was the Wavestation. Why, then, did so many players feel the need to get stuck on Ski Jam, the very first - and utterly identifiable - patch?

Mind you, a few may have jabbed the right-most cursor button a couple of times, landing on The Wave Song, a nearly identical demonstration of the instrument's powerful wavesequencer, but it amounts to the same curious lack of imagination.

Admittedly, programming the Wavestation could be an exercise in frustration, and the sound is indeed very cool, but this particular rhythm was ubiquitous in the early 1990s, finding its way onto loads of pop songs, soundtracks, and seemingly every local broadcaster's station ID.

10. The Super Saw (from the Roland JP-8000)

Virtual analogue synthesizers aren't exactly thin on the ground these days, but once upon a time they were a new and exciting rarity, promising an idealised version of classic analogue synthesis in a modern, stable and dependable package.

Roland's JP-8000 was one of the very first such instruments to hit the scene. While paying homage to instruments of the past, the JP-8000 included many modern niceties such as the ability to remember knob and slider movements and the soon-to-be overused Super Saw waveform, which consisted of seven detuned sawtooth waves played back from a single oscillator.

A much-imitated sound, Super Saw patches would become a favourite of trance producers.

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The 10 best synth presets in history (and where you can hear them)

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Preset synth patches are either a boon or a bane, depending on one's programming prowess. For many 'serious' synthesists, they represent something of a cop-out - easy selections of spoon-fed solutions to impress the inebriated denizens of the local Holiday Inn lounge. However, the vast majority of keyboard players depend on presets to compose and record their arrangements.

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Cambridge-mt

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The audio comparison files in this library are specifically designed to demonstrate the effects of microphone positioning as clearly as possible. To this end, each set of comparison files is made up of performances which have been multimiked using microphones of the same model, thereby minimising differences between the files on account of performance variation and microphone design. Because of this, you can import each family of files into your DAW and mix them together if you wish, perhaps to experiment with multimic combinations or phase interactions. All files are presented both in MP3 format (with an in-browser preview button) and in uncompressed 16-bit/44.1kHz WAV format.

Below you can find links to the instrument-recordings available so far. Like the 'Mixing Secrets' Free Multitrack Download Library, this will be a continually expanding resource, so if you want to keep track of the latest uploads please subscribe to my monthly spam-free Small-studio Secrets News mailing list. I usually post to this list towards the end of the month, and it also includes information about the latest additions to my Mixing Secrets For The Small Studio resources. If you have any suitable multimic recordings that you'd be willing to contribute to the library, then do get in touch.

Many musicians, engineers, and companies have generously helped with the recording sessions that generated these audio comparison files, and I've credited them individually on the specific instrument pages. If you find these pages useful, then please follow the links and show your appreciation in any way you can. All downloads from this site are provided free of charge for educational purposes only, and the material contained in them should not be used for any commercial purpose without the express permission of the copyright holders.

A range of both plucked and bowed acoustic-bass performance techniques were captured through two different eight-mic arrays, each made up of five identical small-diaphragm condenser mics and three identical ribbon mics. A DI signal was also taken from the instrument's bridge pickup so you can experiment with combining that with the mic positions. (90 files)

Three guitars, two multimic setups, and 22 mic positions, as well as demonstrations of absorption/reflection measures and some stereo mic combinations. (96 files)

A seven-foot Yamaha grand piano recorded in a medium-sized rectangular recital hall. A total of nine multimic setups were recorded, both inside and outside the piano, and with variations for half-open, closed, and removed lid. A variety of different stereo configurations are also demonstrated. (73 files)

A Yamaha upright piano recorded in stereo in a medium-sized rectangular recital hall. Different multimic setups were recorded with the instrument's panels on and off, and variations were also done for three different miking distances. (18 files)

Two different clarinets (a Bb clarinet and a bass clarinet) were captured through two different five-mic arrays within a typical home-studio overdubbing environment. (20 files)

Four different cymbals (two crashes, a ride, and a china) were captured through two different five-mic arrays. (40 files)

A Fender Telecaster was played through a rather natty snakeskin-finish Fender Twin Reverb amplifier, and recorded using four different multimic rigs. (11 files )

A western flute was captured through two different five-mic arrays within a typical home-studio overdubbing environment. (10 files)

A single kick drum was captured through a multimic array of five mics as part of a full-drumkit performance. Several additional mics were also recorded alongside to allow evaluation of the mic positions within the context of common 'inside+outside' and 'front+back' multimiking configurations. Two well-known spill-reduction methods are showcased too. (76 files)

Four different saxophones (soprano, alto, tenor, and baritone) were captured through two different five-mic arrays within a typical home-studio overdubbing environment. (40 files)

A snare drum was captured through a multimic array of five mics as part of a full-drumkit performance. In addition, three different snare drums were recorded with the same above/below mic close-mic pair. All files are provided with overhead-mic recordings alongside for context. (17 files)

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The audio comparison files in this library are specifically designed to demonstrate the effects of microphone positioning as clearly as possible. To this end, each set of comparison files is made up of performances which have been multimiked using microphones of the same model, thereby minimising differences between the files on account of performance variation and microphone design.

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Could this neat but powerful little trick save you time and frustration on each and every mix?

Mix engineers who specialise in mainstream pop, EDM or hip-hop are often presented with projects involving high track counts. This is often due to the extensive layering of sounds, shaped with heavy band-pass filtering, that plays such a huge part in the overall 'colour' of tracks in these genres. But even rock and indie projects from traditional four or five-piece bands can still leave you confronted by what feels like a million layered parts, with stacks of doubled and overdubbed guitar parts, multiple mics on everything, huge vocal arrangements and so on. If you're not working at the level where you can afford the luxury of an assistant to weed through the mess and set the mix session up for you, at best this means a lot of work and time spent on setting up the project. It might even mean turning down a project. Picture the scene...

Artist: "Hey Ed, how's it going?"

Me: "Mmm. What's up?"

Artist: "Nothing... Um... you know that EDM track you said you'd mix for us?"

Me: "Yep."

Artist: "I've uploaded it to the server..."

Me: "Server? What's wrong with the Dropbox?"

Artist: "Er, the files are a wee bit too large."

Me: "Too large? We discussed this for a whole hour, didn't we? Anything above 60 tracks will take me at least a week."

Artist: [silence]

Me: "Do I need to book a hitman?"

Artist: [more silence]

Me: [sigh] "OK... Exactly how many tracks are we talking?"

Him: "Um... a shade over 160."

[Dead ring tone]

OK. Nobody wants to go for the nuclear option, particularly if it's a promising track and artist. So how do you make sense of it all before you can begin to mix? The first and foremost process for any engineer to perform is what I call a 'level and pan mix': a stage during which you establish the initial gains for each source and create some basic separation through panning. You end up with a kind of rough, static balance that, later, you can go on to refine with dynamics processing, automation, mutes and all your own neat little mixing tricks.

The usual approach is to choose one sound to act as your main level reference, and balance all the other tracks against it. A lot of people start a mix by setting a level for the kick drum, snare and bass, for example, reasoning both that these will form the rhythmic spine of a track (and so are worth focusing on early) and that, as they're energy-rich sounds, they'll register higher on the meters than most sources - so setting them to an appropriate level reduces the risk of 'overcooking' things at the master bus as the mix progresses. Others might choose to focus on the lead vocal if that's going to be the star of the show, and build the mix around that. And that's all perfectly valid...

But there is another way, which is quick, pretty easy and, surprisingly, fairly accurate: instead of referencing a source from within your mix, you use a pink-noise generator to provide your level reference and balance each and every track to that, one by one. It might sound like crazy pseudo-science but it works - and in this article I'll tell you why and then prove it.

I've already said that you should use pink noise as your reference, but what makes pink noise so special? Why not white noise, for example? After all, white noise is the one with a flat response, right? Well, yes, it's true that white noise has constant energy right across the frequency spectrum. But while this might make it useful for measuring audio equipment, it means it's of naff-all use for mixing. Why? The human hearing system's frequency response is not linear but logarithmic. In other words, we judge pitch increases by octaves, not by equal increments of frequency; each successive octave spans twice as many Hertz as the previous one down the scale. And this means that when we listen to white noise, it appears to us to increase in level by 3dB per octave.

A more 'human-friendly' distribution of energy results when we filter white noise so that it loses 3dB per octave (going upwards). Each octave then contains the same amount of energy as the next one - so it sounds naturally 'balanced'. Well, this is precisely what pink noise is.

White and pink aren't the only types of noise, but you needn't worry about the other colours in this context. (If you're feeling a nerdy need to fill your head with more information on noise types, the Wikipedia page about 'Colors of noise' is as good a place to start as any.)

So, how exactly do we make this pink noise work to our advantage when mixing? The idea is that you use a noise generator (I'll recommend a free one in a moment) to create the pink noise, and calibrate it so that, at the stereo bus, it registers a sensible average level for mixing on your meters. With the noise level set, you solo your first source, so that it alone plays alongside the pink noise, and balance it directly against the noise by ear. You're aiming to find the level at which the source is just barely audible above the noise, but not hidden. Then mute that source, solo the next one and repeat. And go through each and every source in the same way. Take away the noise and you'll have your basic level mix. That's just an overview, though. Now I'll take you through things step-by-step.

First, you need a noise generator plug-in. The freeware MDA TestTone ( http://mda.smartelectronix.com/effects.htm) is ideal because it allows you to pass the incoming audio signal through unaffected - and this means it can live in an insert slot on your master stereo bus, avoiding the need to set up more complex routing, or to insert noise generators on multiple tracks. Furthermore, it can live, bypassed, on the stereo bus for the whole project and act as a useful sanity check for your mix balance later on (if anything's leaping out very far above the noise, it's a sign that it may require your attention). The whole MDA plug-in suite, which includes some other useful freebies, is available for Windows and OS X but it comes in VST format only. That's fine for most DAWs, but Logic and Pro Tools users, for example, will need to find a wrapper of some sort or a suitable alternative noise generator - the basic principles remain the same, though, so you'll still be able to follow the example.

I hope you already know that you should be leaving ample headroom on your master bus when mixing (you can add make-up gain and perform any desired loudness processing later if you need to, but for now we're interested in maintaining headroom and creating a nice mix with a good dynamic range). So before we start riding the faders and generally enjoying ourselves, we need to calibrate the noise so it provides a sensible reference level to allow for this.

We're really concerned with the RMS (average) level, rather than the peak. Peak metering has its uses, but we don't need to worry about peak transients tripping compressors and so on. Some DAWs have RMS meters on the master bus, some don't, so you may need some sort of meter plug-in. I use Cubase, which doesn't display RMS on the main stereo bus channel but has a dedicated master channel which can display all sorts of useful information. To keep things simple for users of other DAWs, though, I used Meterplugs' K-Meter (reviewed in SOS October 2014: http://sosm.ag/kmeter) in the example. Each channel has both peak and RMS meters, and I can recommend this plug-in for its sheer simplicity and elegance.

If you want to follow step by step, place TestTone in an insert slot of your stereo bus, and K-Meter in the following one. Set TestTone's Mode slider to PINK, to generate pink noise, and make sure the Channel is set to CENTRE (so that it outputs in both the L and R channels - it should be on this setting by default). Leave F1, F2, Sweep and '0dB =' alone.

Now it's time for you to set your project's reference level. Use TestTone's Level slider (I've found that there's a bug whereby if you type in the text field it doesn't work) to set the noise to a level that's not too loud, using the meter. In my example, I opted for the K-14 metering scale on K-Meter. This places the zero 14dB below digital full scale - in other words, there's a whopping 14dB above your zero 'target', which guarantees that there's acres of headroom when mixing. With this system in place, you don't have to worry about what all those individual channel meters are doing, or about overloading the mix bus while mixing. You're aiming for the pink noise to generate a sensible meter reading that sits well within the headroom. I found that setting TestTone's pink noise source to -11dB gave me an RMS level of somewhere around -9dB on the K-14 scale, with the peaks hovering around the zero mark.

Finally, as well as the noise itself, you're going to need to be able to hear the audio from your source tracks, so make sure TestTone's Thru parameter is set to 0dB, which is the maximum setting, with the slider fully to the right. (If Thru is off then you won't hear any of the sources; if you alter the Thru level you'll need to go back and adjust the noise level.)

I'll take you through the rest of the process by describing how I used it for an R&B track called 'Are You Ready', by GAM Productions. It has a relatively modest 28 tracks to work with, but even then it took me less time to conduct my level mix that to read this far into the article.

Just as when setting up any mix, before tweaking any track levels I did a bit of housekeeping. I grouped and colour-coded all the channels to make project navigation easier. The groups can help if you're working with multi-miked sources too, as when you've balanced your mic signals, you can simply solo the group instead of each individual track. If you know that things are going to live mostly on the left or right or in the centre of your mix, you can set the basic pans at this stage too.

I left all of the channel faders at unity gain, as this makes the later stages of mixing (automation in particular) easier. Instead, I used Cubase's built-in channel input gain function to balance each individual channel/sound against the noise. If your DAW doesn't offer channel gain, you can insert a gain plug-in in the first insert slot (Blue Cat's Gain Suite is a good cross-platform freebie), or if you're dealing with continuous audio, you can use the clip volume on the arrange page.

I also set up my project for 'solo exclusive' operation, whereby when you solo one track it disables any other solos in your project. This isn't essential, but it does help to prevent errors. After all, I'm only interested in mixing one channel/group at a time against the noise; I don't need to hear any additional channels.

The next step was to start setting the channel levels. I took each channel in turn, and adjusted its gain until I could just about hear the source poking through the noise. The process is that simple: if it sounds a little louder than the noise then back off the level, if you can't hear it at all, increase the gain.

The one complication is that certain frequencies and types of sound will cut through the pink noise just a little better than others, so you'll need to take into account each sound's characteristics when mixing to noise - this becomes instinctive with a tiny bit of practice. A good example of a problematic sound is a sub kick. Quite often, the low frequencies seem masked by the noise and so it seems tempting to boost the channel gain. You need to fight that temptation: as long as the kick is audible, the process will take care of the rest. The same problem occurs with high-frequency sounds, hi-hats being a prime example. Balance with this in mind.

I also often find that it seems to work best if you leave the send effects (like reverbs and delay) muted while performing this pink-noise level mix, and add them in later to finesse things as you mix in earnest. If in doubt, while you're trying this technique, attempt it both with and without your send effects and see what you think.

Finally, the area where many people trying this technique seem to get stuck is vocals. The problem is that vocals are quite dynamic: even the most consistent performance will have louder and quieter sounds within words and phrases. The trick is to focus your ears not on the 'attack' phase of the vocal sound, where consonants and more pronounced note-onsets are usually more easily detected, but instead to make sure that the whole vocal line is heard under the noise.

When you've done all this, switch off the pink noise generator (bypass the TestTone plug-in) and listen to the balance. It won't be perfect - you might want to nudge a couple of faders for any problem sounds, and you'll still have to do all your processing, automation and other mix tricks. But you should find that it has put you quite a way along the right road very quickly.

Now you can get on with the fun part of mixing - and if you manage your channel gain structure correctly while processing your individual sources (EQs with a constant-loudness function such as Tokyo Dawn Slick EQ and lkjb's Luftikus can be a real help here) then you should find that you need to do even less tweaking of your channel faders than normal.

Finally, if you find yourself benefitting from this approach, but that the overall tonal balance that results from mixing to pink noise is not quite to your taste, it's worth bearing in mind a couple of things. First, you can use it to balance layered parts, such as all those stacks of backing vocals within their own group, without applying it to the whole mix. But second, you can get a little more creative with the technique. In particular, you can EQ the pink noise to give a different response to work to. You can even use a 'match EQ' such as Voxengo's Curve EQ (bundled with Cubase 7 and onwards, but available commercially to other DAW users) to capture an average frequency response from a track that you like, and use this to shape your pink noise reference. But that's something to explore in more detail another time...

I've placed an audio example of the track I used to demonstrate the pink-noise technique on the SOS web site at the address given below. I conducted a basic level mix and nothing else - no pans, no effects or dynamics processing, but it took me only four minutes!
sosm.ag/pinknoise-media

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Could this neat but powerful little trick save you time and frustration on each and every mix? Mix engineers who specialise in mainstream pop, EDM or hip-hop are often presented with projects involving high track counts.

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Dennis DeSantis

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Creative Strategies for Electronic Music Producers | Making Music book by Ableton

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A collection of solutions to common roadblocks in the creative process, with emphasis on solving musical problems, making progress and finishing what you start.

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SoundWorks Collection: The Sound of Blade Runner 2049

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In this exclusive video profile we feature the sound team behind Director Denis Villeneuve's, Blade Runner 2049 featuring supervising sound editor Mark Mangini, sound designer Theo Green, re-recording mixer Doug Hemphill and Ron Bartlett, picture editor Joe Walker and Composers Hans Zimmer and Benjamin Wallfisch.

Officer K (Ryan Gosling), a new blade runner for the Los Angeles Police Department, unearths a long-buried secret that has the potential to plunge what's left of society into chaos. His discovery leads him on a quest to find Rick Deckard (Harrison Ford), a former blade runner who's been missing for 30 years.

For more exclusive videos check out www.SoundWorksCollection.com
Join the SoundWorks Collection Vimeo Channel: vimeo.com/channels/soundworkscollection

NOTE: Feel free to re-embed and share this SoundWorks Collection video, but please do not download source video and repost without permission.

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Simon Cann

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How To Make A Noise: a Comprehensive Guide to Synthesizer Programming is perhaps the most widely ready book about synthesizer sound programming. It is a comprehensive, practical guide to sound design and synthesizer programming techniques using:

subtractive (analog) synthesis

frequency modulation synthesis (including phase modulation and ring modulation)

additive synthesis

wave-sequencing

sample-based synthesis

Get the Book

The book and the accompanying patches are available as a free download.

Click here to get the download.

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How to Make a Noise: a Comprehensive Guide to Synthesizer Programming by Simon Cann

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How To Make A Noise: a Comprehensive Guide to Synthesizer Programming is perhaps the most widely ready book about synthesizer sound programming. It is a comprehensive, practical guide to sound design and synthesizer programming techniques using: subtractive (analog) synthesis frequency modulation synthesis (including phase modulation and ring modulation) additive synthesis wave-sequencing sample-based synthesis The book and the accompanying patches are available as a free download.

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Interactive demos

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The ring modulator is best known as the device used to create the terrifying voices of the Daleks and the Cybermen in Dr. Who.

The ring modulator works by taking two inputs and multiplying them together. In the original Ring Modulator a tape loop with a 30Hz sine wave tone was combined in real time as an actor spoke into a microphone. The diodes in the machine also gave the effect its characteristic distortion.

An early piece of equipment, the ring modulator is still in use today.

Have a play with our simulation of the Ring Modulator. Adjust the speed of the tape loop to change the effect, then keep scrolling to find out how we built it using the Web Audio API.

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The ring modulator is best known as the device used to create the terrifying voices of the Daleks and the Cybermen in Dr. Who. The ring modulator works by taking two inputs and multiplying them together. In the original Ring Modulator a tape loop with a 30Hz sine wave tone was combined in real time as an actor spoke into a microphone.

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Discussion

synthesis and analysis

The distinction between music and noise is mathematical form. Music is ordered sound. Noise is disordered sound.

Music and noise are both mixtures of sound waves of different frequencies. The component frequencies of music are discrete (separable) and rational (their ratios form simple fractions) with a discernible dominant frequency. The component frequencies of noise are continuous (every frequency will be present over some range) and random (described by a probability distribution) with no discernible dominant frequency.

Sound is a longitudinal wave, which means the particles of the medium vibrate parallel to the direction of propagation of the wave. A sound wave coming out of a musical instrument, loudspeaker, or someone's mouth pushes the air forward and backward as the sound propagates outward. This has the effect of squeezing and pulling on the air, changing its pressure only slightly. These pressure variations can be detected by the ear drum (a light flexible membrane) in the middle ear, translated into neural impulses in the inner ear, and sent on to the brain for processing. They can also be detected by the diaphragm of a microphone (a light, flexible membrane), translated into an electrical signal by any one of several electromechanical means, and sent on to a computer for processing. The processing done in the brain is very sophisticated, but the processing done by a computer is relatively simple The pressure variations of a sound wave are changed into voltage variations in the microphone, which are sampled periodically and rapidly by a computer and then saved as numbers.

A graph of microphone voltage vs. time (called a waveform) is a convenient way to use a computer to see sound. Before the rise of ubiquitous digital computers, waveforms were often analyzed electronically using an oscilloscope - a cathode ray tube with an electron beam that traced voltage as a function of time on a fluorescent glass screen. Oscilloscopes are basically simplified televisions with one purpose (to a draw time series or parametric graph) and one color (usually bright green). This task is easily mimicked by 21st century desktop, laptop, and tablet computers as well as smart phones. Oscilloscope applications on these devices often pay homage to their analog ancestors by using a green color scheme.

The simplest sound to analyze mathematically is the pure tone - one where the pressure variation is described by a single frequency. A pure tone would look like a sine curve when graphed oscilloscope style.

y = A sin 2π ft

where...

y =

the instantaneous value of the microphone voltage (∆ V), which is directly proportional to the variation in air pressure ( P) due to sound waves impacting the microphone

2π =

a constant needed to get the units to work out

040 Hz pure tone

Music in its simplest form is monotonic; that is, composed only of pure tones. Monotonic music is dull and lifeless like a 1990s ringtone (worse than that even); like a 1970s digital watch alarm (now we're talking); like an oscillating circuit attached to a speaker built by a college student in an introductory physics class (so primitive). Real music, however, is polytonic - a mixture of pure tones played together in a manner that sounds harmonious. A sound composed of multiple frequencies like that produced by a musical instrument or the human voice would still be periodic, but would be more complex than just a simple sine curve.

The human voice and musical instruments produce sounds by vibration. What vibrates determines the type of instrument.

category

vibrating part

examples

idiophone

whole
instrument

bell, cymbal, musical saw,
wood block, xylophone

membranophone

stretched
membrane

drums, kazoo, human voice

chordophone

stretched
string

strings (violin, guitar, harp), piano

aerophone

air
column

woodwinds (saxophone, flute),
brass (trumpet, tuba), organ

electrophone

electric
circuit

synthesizer, theremin

Like many other mechanical systems, musical instruments vibrate naturally at several related frequencies called harmonics. The lowest frequency of vibration, which is also usually the loudest, is called the fundamental. The higher frequency harmonics are called overtones. The human auditory system perceives the fundamental frequency of a musical note as the characteristic pitch of that note. The amplitudes of the overtones relative to the fundamental give the note its quality or timbre - pronounced in English as tæmbər or in quasi-French by English speakers as tɛ̃br with a nasal ɛ̃ for the medial e and silence for the final e. Timbre is one of the features of sound that enables us to distinguish a flute from a violin and a tuba from a timpani.

Recall that when waves meet they don't collide like material objects, they pass through each other like spectres - they interfere. Interfering waves combine by the principle of linear superposition - basically, just add the values of one function to the values of another function everywhere in mathematical space. With the right combination of sine and/or cosine functions, you can make functions with all kinds of shapes (as long as they're functions). Several examples were illustrated earlier in this book. Here they are again...

square wave

y = ∑

1

sin(2 n − 1) x

2 n − 1

sawtooth wave

y = ∑

(− 1)n +1

sin nx

n

triangle wave

y = ∑

1

cos(2 n − 1) x

(2 n − 1) 2

The act of combining pure tones together to produce a complex waveform is called additive synthesis. Non-electronic instruments do this naturally. Electronic instruments (if they want to be taken seriously) are designed with additive synthesis in mind. The analog synthesizer is a good example of this.

Generate a sinusoidal electric signal of with an oscillator - a fairly simple electric circuit composed of a capacitor (C) and an inductor (L), also known as an LC circuit. Using a second oscillator generate a second signal at a different frequency. The second harmonic would be a wise choice. Add a third, fourth, fifth, sixth and seventh oscillator for the corresponding higher harmonics. Synchronize them electronically so that when the pitch of the fundamental oscillator changes, the frequencies of the overtone oscillators follow along. Adjust the relative amplitude of each oscillator to produce a sound with the desired timbre. Attach the circuit to an amplifier and a loudspeaker so it can be heard. Attach the fundamental oscillator to a control device that turns it on and off and changes its frequency as needed. Make your control device look like a piano keyboard so musicians will use your device as a musical instrument. Call it a synthesizer since it uses the principles of additive synthesis. Let the Seventies begin.

It is mathematically possible to transform any waveform from a continuous sequence of values on a time-series graph into an infinite series of discrete sine and cosine functions. The process is called spectral analysis or Fourier analysis after its inventor, the French mathematician and physicist, Joseph Fourier (1768-1830). Taking a sound apart through spectral analysis is the inverse process of putting a sound together through additive synthesis.

Start with an arbitrary periodic function and assume that it was made from the linear superposition of an infinite number of sine and cosine functions in a harmonic series. What coefficients should each harmonic be multiplied by to give the desired function? The process for answering this question is called a Fourier transform and it is best left to a mathematician to explain.

Any periodic function f( T can be written like this... t) with a period of

f( t) = a 0 + ∑ an cos

⎛
⎜
⎝

2π nt

⎞
⎟
⎠

+ ∑ bn sin

⎛
⎜
⎝

2π nt

⎞
⎟
⎠

T T

where...

T

a 0 =

1

⌠
⌡

f( t) dt

T

T

an =

2

⌠
⌡

f( t)cos

⎛
⎜
⎝

2π nt

⎞
⎟
⎠

dt

T T

T

bn =

2

⌠
⌡

f( t)sin

⎛
⎜
⎝

2π nt

⎞
⎟
⎠

dt

T T

...and n = 1, 2, 3,... and that's all I want to say about that.

Fortunately for us, relatively simple algorithms for solving this problem were discovered soon after Fourier invented the technique, and computer scientists have been coding and sharing fast Fourier transforms since the 1960s. Audio applications with FFT are easily found online for many computing platforms - including smart phones, which means that sound spectral analysis is now literally with everyone's grasp.

The graph above shows the spectral analysis of a female voice. Sounds of all sorts of different frequencies are being produced - some more intensely than others. It appears that this particular voice was strongest at 270 Hz (C♯ 4 for those of you who know music) and multiples of 270 Hz (540 Hz, 810 Hz, 1080 Hz, 1350 Hz, 1620 Hz, and 1890 Hz). This graph has been normalized, which exaggerated the peaks at the higher frequencies so they could be seen. I like this graph because the spacing between peaks is just so perfect. This kind of structure is what makes a sound musical. Let's look at some more examples.

A flute is essentially a tube that is open at both ends. Air is blown across one end and sound comes out the other. A spectral analysis confirms this. The harmonics are all whole number multiples of the fundamental frequency (436 Hz, a slightly flat A 4 - a bit lower in frequency than is normally acceptable). Note how the second harmonic is nearly as intense as the fundamental. This strong second harmonic is part of what makes a flute sound like a flute.

A recorder is also a tube with two open ends. It produces a sound similar to a flute, but not exactly the same. Again the harmonics are whole number multiples of the fundamental frequency (923 Hz, a very sharp A 5 - much higher in frequency than is normally acceptable), but for some reason the second harmonic is nearly non existent. This nearly missing second harmonic is part of what makes a recorder sound like a recorder and not sound like a flute.

A tuning fork is forked; that is to say, it splits from its handle into two branches called tines. Each tine is fixed to the handle at one end, but is free to vibrate at the other. As a result, one would expect to find only those harmonics that were odd multiples of the fundamental in the spectrum of a tuning fork. This is what the spectral analysis shows. The even harmonics are present, but are they are weak and are probably due to the sympathetic vibrations of something nearby. This spectra was produced by striking a large, demonstration-size tuning fork (not the one pictured above) with an excessively heavy blow. Tuning forks should always be tapped lightly and on a resilient surface. Doing so reduces the intensity of the "ping" overtones, which is a desirable thing. An ideal tuning fork would vibrate at just one frequency. The tuning fork used in this experiment was rated at 256 Hz, but the spectral analysis software picked up 259 Hz for the fundamental. Is the tuning fork out of tune or is the software in error?

noise

Music is sound with a discrete structure. Noise is sound with a continuous structure. Music is composed of sounds with a fundamental frequency and overtones. Noise is composed of sounds with frequencies that range continuously in value from as low as you can hear to as high as you can hear - not necessarily at equal intensity, however. Music is described mathematically by an infinite sum of sines and cosines multiplied by appropriately valued coefficients - infinite mathematically, but in practice only a handful of overtones really matter. Noise is described by a spectral power distribution (or power spectrum), much like the statistical distributions of kinetic molecular theory. Music is ordered. Noise is random.

Noise is what you hear when you tune an analog radio or television to an empty frequency. It's the overall sound of rain falling on leaves, soda bubbling in a glass, air escaping from a tire, or a crowd applauding.

"Noisy" describes some voiceless consonants used in English better than "musical".

sound

ipa*

example

name

/hw/

[ʍ]

wh eat

voiceless labiovelar approximant †

Noise might not be periodic, but it can still be analyzed with a fast Fourier transform - as long as we only examine a finite amount of it. Since infinite quantities don't exist in the real world, this isn't a problem. Mathematics allows for more possibilities than physicists will never encounter.

Noise might not be ordered, but that doesn't mean it can't be described. Frequency is to sound as color is to light. We hear different frequencies of sound as different pitches (A, B, C, D, E, F, G; for example). We see different frequencies of light as different colors (red, orange, yellow, green, blue, and violet; for example). The analogy is not perfect. (What analogy is?) The notes of the musical scale repeat themselves for every doubling of frequency (a topic I'll come back to). The frequencies of visible light span a range that is so narrow, they never get a chance to double. The frequencies of the notes of a musical scale are related by simple numerical fractions. The frequency bands associated with specific color names show no mathematical relationships. (This analogy isn't looking very promising.)

In much the same way that most sounds we hear are polytonic (composed of many frequencies), most light we see is polychromatic (composed of many frequencies). Sources of light that are hot enough (like the sun) will emit a mixture of frequencies that span the entire visible spectrum. The color of this light is white - a mixture of red, orange, yellow, green, blue, violet, and everything in between. White noise is the audio equivalent of white light. It's a combination of all the frequencies that span the entire audible spectrum. White noise is a mixture of all the musical pitches with names and all the pitches in between that aren't named. This is the informal definition of white noise.

Formally, white noise is a sound with a flat power spectrum; that is, it transmits power equally at all frequencies. It is a mathematical ideal with a representation something like this...

p( f) = constant

So that...

fmax

P =

⌠
⌡

p( f) df

fmin

where...

p( f) =

the value of the power spectral density measured in W/Hz (used for theoretical discussions) or the relative power spectral density in V 2/Hz (what normally gets used in practice)

Note...

Spectral analysis applications process the voltage of the signal coming from a microphone, not the power of the sound detected by the microphone. Electric power is proportional to the square of voltage at constant resistance, so V2/Hz are measured instead of W/Hz. Both quantities have the same mathematical structure, which is the important thing in audio analysis.

Frequencies are limited to a definite range. This keeps the integral from running away to infinity. A rectangle of any height that is infinitely wide would contain infinite area. Since we can't let this happen, we have to assign limits to the spectrum. This is something that makes the white noise distribution unrealistic. Real statistical distributions tend to be curves that taper off nicely to zero. They don't start and end abruptly.

White noise is a simple mathematical model that is so simple, it's unrealistic. It is a good way to introduce the topic of power spectral density, however. It is also a convenient reference sound for testing audio equipment.

The human auditory system is logarithmic when it comes to perceiving sounds. This is true when it comes to amplitude (which is related to intensity or loudness and was discussed earlier in this book) and frequency (which will be discussed now). Listen to a sequence of two pure tones with a difference of 100 Hz between them, say 100 Hz and 200 Hz. Then listen to another sequence with the same linear difference but starting on a much higher frequency, say 800 Hz and 900 Hz. The perceived increase in frequency of the second pair seems smaller than the first pair even though the absolute size of the increase is the same.

A doubling from 100 Hz to 200 Hz sounds similar to a doubling from 800 Hz to 1,600 Hz.

Likewise a one eighth increase from 100 Hz to 112.5 Hz sounds the same as a one eighth increase from 800 Hz to 900 Hz.

We tend to group sounds perceptually into bands that increase by successive powers of 2. A mathematician would call this a logarithmic increase. Musicians call the bands octaves. Audio technicians developed pink noise as a response to this reality.

Formally, 1/ pink noise is a sound with a power spectral density that is inversely proportional to frequency over the range of audible frequencies. Because of this it is also known as f noise (one-over-f noise). We can write this as a proportionality statement...

...or as an equation with a constant of porportionality ( k)...

Following the analogy that sound is to frequency as light is to color, pink noise is pink because it's a little bit white and a little bit red - white since all frequencies are present, red since the lower frequencies are the louder ones. (Low frequency light appears red.)

Like white noise, a pink noise source cannot span an infinite range of frequencies. If it did, it would be radiating infinite power.

∞

∞

P =

⌠
⌡
k
df = k

⎡
⎢
⎣

log f

⎤
⎥
⎦

0

0

A pink noise signal transmits energy equally over all octaves (logarithmic intervals). Let's test this essential characteristic. Compute the power radiated in a frequency band spanning the octave f = ( a, 2 a).

P =

⌠
⌡
k
df = k

⎡
⎢
⎣

log f

⎤
⎥
⎦

P =

k (log 2 a − log a)

P =

Compare that to the power radiated in the next octave higher f = (2 a, 4 a).

P =

⌠
⌡
k
df = k

⎡
⎢
⎣

log f

⎤
⎥
⎦

P =

k (log 4 a − log 2 a)

P =

Listen to the audio samples below to hear what white and pink noise sound like. Pink noise sounds richer than white noise - like a waterfall or blowing wind. White noise sounds brighter, but also more harsh than pink noise - like steam escaping a radiator or, as a famous biologist once said, like " innumerable mice eating Rice Krispies". A cheering crowd has it's own unique power spectrum. It's neither white nor pink.

consonance & dissonance

Sounds that lend themselves to music are those that sound pleasant (or at least tolerable) when played in sequence (called melody) or in unison (called harmony).

high degrees of consonance (shared harmonics)

dissonance

Consonances are sometimes described as being inherently more pleasant to the ear and dissonances as less pleasant.

Musical notes that are consonant share a large number of overtones. In the just intonation scale the octave is the most consonant interval followed by the perfect fifth, perfect fourth, major third and sixth, and major seventh. No other set of intervals shares such a high degree of consonance.

Notes separated by an octave sound similar - like two people with different voices trying to sing the same note.

overtones line up

Helmholtz used the German word oberton which literally means "upper tone". Someone transliterated the word instead of translating it and oberton became overtone in English by accident. If Helmholtz had meant overtone he would have said überton.

Stuff

pure tones

complex tone

partials

fundamental

overtones

harmonic

inharmonic

evil 7th overtone

who cares after the 8th overtone

This is where the physics ends and the music theory and abstract algebra (group theory and combinatorics) start to take over.

musical scales

The foundation of music is the musical note: a combination of pitch (the musical word for frequency) and duration (the ordinary word for amount of time).

Did I say music was based on notes? That's not true. Real music is based on intervals (the ratio of any two pitches) with high degrees of consonance (shared harmonics).

A scale is set of pitches (pitch classes, more precisely) arranged in order of increasing frequency from which notes are selected and arranged to create a musical composition. The exact pitches used in any scale are determined by the starting pitch (called the tonic), a set of rules for generating intervals (called a tuning system), and the pattern of intervals selected (sometimes called the mode).

Western music theorists identified eight basic intervals defined by their relative size...

tonic or unison - an interval of one to one, perfect consonance

second

third

fourth

fifth - the third most consonant interval

sixth

seventh

octave - an interval of two to one, half of all overtones match

The intervals of adjacent pitches in a scale often come in two sizes called whole tones and semitones (or half tones). The relationship between these should be obvious - two semitone intervals applied in succession equal a whole tone - but it turns out to be only approximately true in most tuning systems. Combinations of tones and semitones may also be named by size...

semitone - typically the smallest interval

whole tone - two semitones

ditone - two whole tones (a term that's rarely used)

tritone - three whole tones, half an octave

6 whole tones or 12 semitones equal one octave.

Each interval also needs an adjective to describe its quality...

perfect - an interval that is an inversion of another

major - the larger of two nearly equal intervals

minor - the smaller of two nearly equal intervals

augmented - a semitone higher than a major or perfect interval

diminished - a semitone lower than a minor or perfect interval

Many, many, many interval patterns are used in musical composition. (That's too many manys.) A convenient way to organize them us by the number of intervals they contain. Here's a list of the interval patterns that will be discussed in this book.

pentatonic

hexatonic

heptatonic

octatonic

dodecatonic

Since notes separated by an octave sound similar, a musical scale can be completely described by a set of intervals with ratios ranging from 1:1 to 2:1. If you need notes above the octave, just double the intervals. If you need notes higher than that, double the intervals again. If you need notes lower than the tonic, take all your intervals and halve them. If you need notes lower than that, halve all the intervals again. You get the idea. This is called octave duplication or circularity of pitch.

Pitches separated by an octave form a pitch class. They are always named using an uppercase letter (A, B, C, D, E, F, G) and sometimes include a modifier symbol called an accidental (♯, ♭, ♮).

symbol

name

description

Adding a sharp to a pitch is the same as adding a flat to the next higher pitch (usually). F♯ is the same as G♭, for example. This is known as enharmonic equivalence. Similarly, adding a double sharp or a double flat to a lettered pitch changes the pitch by one letter (usually). C♯♯ is the same as D, and D♭♭ is the same as C. I keep saying "usually" because there are exceptions around the notes B♯, C♭, E♯, F♭. A piano keyboard is a familiar way to display all the pitch classes.

Enharmonic equivalence is not a given. There are some tuning systems where the distinction between F♯ and G♭ is real. That's part of the reason we have two names for the same pitch today. Thankfully, most of us will never need to know what that distinction is. I have read explanations, but I have never understood them. I think most formally trained Western musicians today would probably say the same thing. Fans of Renaissance and Baroque music would be the exception.

Want to know whether you should say A♯ or B♭? Just look at the pitch classes of the scale used to create your musical composition. If you say any letter twice, you've probably done it wrong. For example...

F, G, A, B♭, C, D, E

is correct, but...

F, G, A, A♯, C, D, E

...is wrong because the letter A appeared twice. This is not a perfect rule, but it works for all the diatonic scales - the most used scales in Western music. (Non-Western music theory is beyond the scope of this book. The same goes for microtonal music - music that uses intervals smaller than a semitone)

name

pitch classes

type

just intonation

An interval is the ratio of any two pitches in a scale. Consonance occurs when the overtones of one pitch coincide with the overtones of another. The intervals with the highest degree of consonance are ratios of small whole numbers (like 2, 3, 5) or small powers of these (like 4, 8, 9, 16, 25, 27) or small products of these (like 6, 10, 12, 15, 18, 20). The more prime factors present in a ratio, the less consonant it sounds. A perfect fifth ( 3 15 3·5 135 3·3·3·5 2·2·2·2·2·2·2). A tuning system built on these principals is described as having 128 = 2·2·2) which is much more consonant than a major chromatic semitone ( 8 = 2) is more consonant than a major seventh ( just intonation. The word "just" here implies that the pitches generated are the "right" ones.

Just intonation is a simple way to build a musical scale. It's the natural tuning system people use when they sing together unaccompanied by musical instruments (a cappella). There are many types of scales with just intonation. I will discuss a few of the basic ones. I am showing them on a piano keyboard since it is a familiar instrument, but pianos are not normally tuned this way. (Pianos use equal temperament, which will be described next.)

diatonic scales

The scale shown below is a type of just intonation scale starting on C. Since the third interval ( 5 4 = 1.25) is "large", the scale is major. Music performed in a major key tends to sound bright, happy, or triumphant. This is true whether the scale was constructed using just intonation or some other scheme. This is probably a function of culture more than anything else. (Happiness isn't something physicists go around measuring - although we do discuss brightness from time to time and in a different context.)

Scales like this one are given the name diatonic - from the Greek phrase δια τονικη ( dia tonike), which means "across the tones". The implication is that all the pitches you need are contained in this scale. That's not quite right, however. (More on this later.)

This particular scale is also heptatonic, which means it contains seven intervals: three major whole tones ( 9 8), two 10 9), and two 16 9). The ratios of adjacent pitches in a major scale follow this order... minor whole tones ( semitones (

major whole tone

minor whole tone

semitone

major whole tone

minor whole tone

major whole tone

semitone

There's no reason I have to start playing a scale on C like the diagrams above show. Why not start playing on A as the diagrams below show? This gets you through an octave with intervals that are still fairly nice. Since the third interval ( 6 5 = 1.2) of this scale is "small", the scale is said to be 27 4 3). Since the fourth is a perfect interval, that must make this an 20 ≈ minor. Music performed in a minor key tends to sound dark, sad, moody, or introspective. Why this happens is outside the realm of physics. It also contains a particularly harsh interval that is close to a fourth ( imperfect fourth. It's an interesting example of how being close is not the same thing as being close enough. A perfect fourth sounds pleasant. An imperfect fourth sound harsh.

This scale is also heptatonic and diatonic. The ratios of adjacent pitches in a minor scale follow this order this order...

major whole tone

semitone

major whole tone

minor whole tone

semitone

major whole tone

minor whole tone

In general, a diatonic scale is any scale that can be played on the white keys of a piano. Since there are seven intervals in a diatonic scale, there are seven pitches to start out on on a piano (A, B, C, D, E, F, and G). Starting on C gives you the major diatonic scale. Starting on A gives you a minor diatonic scale. The remaining five scales are used to lesser degrees in Western music past and present - medieval Christian chant being the example most frequently cited. All seven diatonic scales (or modes as they are called) have names corresponding to regions in ancient Greece. The assignment of Greek place name to diatonic mode is completely arbitrary, however. If there ever was any meaning to this naming convention, it's been lost to time.

aeolian*

locrian

ionian†

dorian

phrygian

lydian

mixolydian

pitch

A

B

C

D

E

F

G

chromatic scales

Let's return to the the C major scale. I find that one more interesting because of what happens when you invert the intervals. Instead of moving from the tonic to the pitch, let's move from the pitch to the octave.

To go from C to G on this scale is a jump up of a fifth (multiply by 3 4 3). This means the fourth is the inversion of the fifth. 2). From G to the next C is a jump of a fourth (multiply by

and the fifth is the inversion of the fourth, which sounds like a statement of the obvious.

In general, any pair of intervals that together results in a jump of one octave is called an inversion.

interval × inversion = octave

That means a third and a sixth are an inversion.

and so are a sixth and a third, which should also be obvious except...

Wait a minute. What just happened? The first first time I wrote the third, I wrote 5 6 8 5 3). Welcome to the world of just intonation, where intervals in one direction on a scale do not necessarily equal those in the inverse direction. 5 ≠ 5. Last time I checked, those were different numbers. I did something similar with the sixth ( 4. The second time I wrote

The tonic, fourth, fifth and octave are not affected by inversion, which is why they are called perfect intervals. The fourth and fifth are given this as a title - as in, "a perfect fourth" or "a perfect fifth". For some reason no one says "a perfect tonic". Maybe because it sounds like the name of a cocktail. No one says "a perfect octave" either, but I don't have a joke for that one.

The second, third, sixth, and seventh are made smaller by inversion. An interval used to raise a pitch above the tonic is always larger than the same interval used to take the new pitch to the octave. As is tradition, larger intervals are said to be major and smaller ones minor.

interval

inversion

major seventh

= 1.875

minor second †

= 1.066...

Something interesting happens when you merge the perfect, major, and minor intervals together into a scale. The minor intervals fill in the "spaces" between pitches separated by a whole tone. (The exception to this is the gap between the perfect fourth and the perfect fifth.) The resulting scale has twelve intervals, each one a semitone away from its neighbor. Such a scale is said to be chromatic - from χρωματος ( chromatos), the Greek word for color. Since color is to vision as pitch is to hearing, the metaphor is entirely appropriate. The chromatic scale is more "colorful" than the diatonic scale.

The intervals of the chromatic scale are too numerous to fit nicely on my piano keyboard illustration. Instead, here's a table showing both the intervals with respect to the tonic and the intervals with respect to the preceding pitch in the just intonation chromatic scale.

interval to

... tonic

... predecessor

semitone

tonic

= 1

minor second 1

= 1.066...

= 1.066...

diatonic

major second 2

= 1.125

= 1.054...

major

tritone 3

= 1.422...

= 1.066...

diatonic

A word about semitones. There are several of them. An interval of 16 15 is called a diatonic semitone. It separates the major third from the perfect fourth and the major seventh from the octave in the major diatonic scale. When we merged the perfect and major intervals of the diatonic scale with their inversions (the minor intervals) the diatonic semitone popped up in four new locations - tonic to minor second, major second to minor third, perfect fifth to minor sixth, and major sixth to minor second. It's a popular interval. The chromatic scale has two other semitones - a larger or 135 128 and a smaller or 25 24. major chromatic semitone equal to minor chromatic semitone equal to

size

interval

Combinations of chromatic and diatonic semitones make whole tones - majors make majors and minors make minors. In a sense, the chromatic semitones are inversions of the diatonic semitone over the span of a whole tone.

chromatic
semitone

×

diatonic
semitone

=

whole
tone

Which brings us to the tritone - the bad boy of the chromatic scale, stuck between the beauty and perfection of the fourth and fifth. We know something belongs there (the size of the interval demands it) but what? All the other intervals arose naturally. That should really say "naturally" in scare quotes. How natural is this process? You shall be born by the forced union of the perfect fourth and a semitone... But which one?

Again I ask... Which one? The perfect fifth and the perfect fourth are inversions of one another. The tritone lies between these, which means it would have to be its own inversion. Do any of these satisfy that recommendation? If so that's a good sign this would be our tritone.

Well that's no good. Two of the tritones are inversions of each other, which doesn't do us any good, and the inversion of the third one looks like it just gave us another tritone. We're going about this the wrong way. Instead of testing fractions hoping one will work out, maybe we should use algebra and just go to the answer directly. What number, x, equals itself when divided into 2?

I didn't expect the square root of two.

Dun, dun, duuuuuunnnnn! Nobody expects the square root of two! Its chief weapon is surprise, fear and surprise; two chief weapons; fear, surprise, and irrationality! Uh, among its chief weapons are: fear, surprise, irrationality, and an almost fanatical inability to be written as a fraction! Uh, I'll come in again...

size

type

√2

= 1.414...

ideal tritone 1

= 1.422...

minor diminished fifth 2

After playing with these intervals for what seems like days, I think I'll just go with 64 45 - the value closest to the √2 ideal. It seems to play nicely with the other intervals - or at least is doesn't torture them much.

pentatonic scales

Truth is ever to be found in simplicity, & not in y e multiplicity & confusion of things.
Isaac Newton, ca. 1680

If the chromatic scale is all about multiplicity, the pentatonic scale is all about simplicity.

The semitone is one of the least consonant intervals. The chromatic scale is nothing but semitones it seems. The chromatic scale is the mother of all scales as a result. Semitones are the price you pay for that amount of versatility. Diatonic scales are considered simple, but they all have two semitones in them. That's two opportunities for a little bit of dissonance to sneak in. What does the risk averse musician do? If you fear semitones or just don't like them, you could eliminate them. Music composed without semitones would be always mellow and never harsh. Since one of the purposes of music seems to be to "sooth a savage breast", let's do it. Let's just get rid of all the semitones.

Return to the C major diatonic scale again. (My favorite starting point.) Take the diatonic scale and drop the fourth and seventh pitch classes, the ones adjacent to the semitone intervals. This gives you a scale with five pitches - a pentatonic scale.

A pentatonic scale is like a diatonic scale in that it is made up of large and small intervals. In the diatonic scale, the large intervals are whole tones and the small intervals are semitones. In the pentatonic scale, the large intervals are thirds (minor thirds for this example) and the small intervals are whole tones. The ratios of adjacent pitches on this pentatonic scale follow this order...

major whole tone

minor whole tone

minor third

minor whole tone

minor third

This kind of pentatonic scale was built using the white keys of the piano. Starting only on the white keys, there are five allowed pentatonic modes. The modes are identified by the location of the missing intervals (the ones that were semitones in the diatonic scale).

minor
pentatonic

major
pentatonic

egyptian

minor
blues

major
blues

second

omit

omit

third

omit

omit

fourth

omit

sixth

omit

omit

seventh

omit

omit

The black keys can also be used to make a pentatonic scale.

This gives a slightly different ratio of adjacent pitches and one minor third that's a little bit flatter than normal ( 32 6 5 = 1.2). It still has the same overall composition of three whole tones and two minor thirds. 27 = 1.185... is a little bit smaller than

major whole tone

small minor third

major whole tone

minor whole tone

minor third

Starting on the black keys, there are five pentatonic modes. Again, the modes are identified by the locations of the missing intervals.

minor
blues

major
blues

minor
pentatonic

major
pentatonic

egyptian

second

omit

omit

third

omit

omit

fifth

omit

sixth

omit

omit

On a piano: the black keys are pentatonic, the white keys are diatonic, and all the keys are chromatic.

equal temperament

I need to write this part.

Less consonant (full of beats), but easier in terms of transposition.

roughening of chords played, increases musical versatility

The chromatic circle is a geometric way to display all the pitch classes. This way is sometimes better for showing the enharmonic relationships. The space between adjacent pitch classes is a semitone.

major

minor

How about an octatonic, "string of pearls" scale?

Whole tone scale

pitch
class

octave

0

1

2

3

4

5

6

7

8

pythagoras' circle of fifths

Pythagoras of Samos was the first to try and describe music with a mathematical system called the circle of fifths (or cycle of fifths). Start with the tonic. Multiply by a perfect fifth ( 3 1 2) so you can keep building your scale. Well... 2). Do it again. This puts you into the next octave. Bring it down an octave (multiply by

3 1 3 4 2 = 2 ×

is the inverse of 4 4 4 3 2 are called the perfect fourth and perfect fifth, respectively. 3 and 3 turns out to be the fourth interval in the series of eight that make up an octave. Thus the name fourth. The fifth and the fourth are inversions of one another in an octave. They are the only intervals that work out this way. That makes them special, in my mind, but the adjective that was ascribed to them was perfect. Thus the intervals 3, an interval with a great deal of consonance. When you completely build the scale, the ratio

So here's the plan again: start with the tonic ( 1 3 1 2 2 1). 1), and repeat until the ratio equals an octave ( 2) when the product excedes an octave ( 2) over and over again, take it down an octave ( 1), bring it up a perfect fifth (

We'll start on C since that's the middle of the modern piano. Behold!

interval

fractional value

decimal value

Oh oh. For those of you familiar with the piano, you will note that the errors occur at notes that do not exist on the keyboard. There is no such thing as a B♯ (A♭) or an E♯ (F♭).

interval

decimal value

interval name

It should really be called the spiral of fifths since it never closes up. One complete lap around the circle should equal a whole number of octaves. That turns out to be 12 fifths and 7 octaves. But 12 fifths is a bit larger than seven octaves. This discrepancy is known as the pythagorean comma and is equal to...

B♯:C = ( 3 2) 1 2) 531,441 524,288 = 1.0136432647705... 12( 7 =

Thus B♯ is a bit higher than C by about one-quarter of a semitone. This is a small difference that would be audible to trained ears were the two notes to be played in succession. Ordinary folks might not perceive the difference at all. Play them together as a part of a chord and your ears would not enjoy it.

f beat = f B♯ − f C
f beat = (256 Hz)(1.013643264771 − 1)
f beat = 3.5 Hz

The dissonance would be audible as a 3.5 Hz beat for C = 256 Hz. No musician would ever want to be this far out of tune and no audience would want to listen to them.

# data[].attributes.attachment_link.title
Music and Noise - The Physics Hypertextbook

# data[].attributes.attachment_link.description
The distinction between music and noise is mathematical form. Music is ordered sound. Noise is disordered sound. Music and noise are both mixtures of sound waves of different frequencies. The component frequencies of music are discrete (separable) and rational (their ratios form simple fractions) with a discernible dominant frequency.

# data[].attributes.body
Vo l2: Harmonic Catalog (sic)

# data[].attributes.headline
Theory and Practice of Musical Sound Synthesis

# data[].attributes.subject
Theory and Practice of Musical Sound Synthesis

# data[].attributes.attachment_link.description
Oscillations and Simple Harmonic Motion. Waves and sound with animations and video film clips. Physclips provides multimedia education in introductory physics at different levels. Modules may be used by teachers, while students may use the whole package for self instruction or for reference.

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a couple dozen short films about PRESETS

# data[].attributes.body
[Computer Music]

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(Image credit: Fair dealing)
Digital audio workstations - or DAWs - are now the mainstay of most producers' workflow, but despite their cutting-edge credentials, you might be surprised at just how far back their roots can be traced. In the early 1980s, the first, primitive versions of what we now call digital audio workstations began changing the way musicians could express themselves.
The musical landscape looked very different back then, with computers just starting to become a viable option for making music, and the introduction of MIDI was a revolution when it came to controlling synths, drum machines and samplers. The idea of recording or editing digital audio on a computer was still a long way off, but even the earliest music programs offered new options that had never existed before.

Why does this matter? To many of today's musicians, what happened more than three decades ago might seem irrelevant to the way we make music today. We wouldn't go so far as to insist you must learn your history to understand how good we have it today, but it really puts into perspective just how far we've come in such a short space of time. Asking a few basic questions can help us understand the incredible music-making power we have at our disposal in software today. What was a sequencer, and how did that evolve into a DAW? How important was MIDI? What about trackers, chiptune and the demoscene? How did newer DAWs change the game? What about the computer platforms and programs that didn't make it?

Join us, then, as we take a nostalgic look back at the origins of some of the most popular DAWs and tell the story of how computer music was born.

As Bob Marley put it: "In this bright future, you can't forget your past."

The godfather of DAWs: Steinberg Cubase

How far back can we trace the story of modern DAWs? At least as far as 1983, when keyboard player Manfred Rürup met engineer Karl 'Charlie' Steinberg and conceived the idea of a software MIDI sequencer. They were truly ahead of the game: the MIDI specification had only been formally announced the same year.

The duo founded Steinberg Research and developed their first program, Multitrack Recorder, a 16-track sequencer for the Commodore 64. Despite selling fewer than 50 copies, the program laid the foundation for a dynasty that continues to this day. Multitrack Recorder became Pro-16, then Pro-24, Cubit and eventually Cubase. Commodore 64 development shifted to the Atari 520ST, then the Commodore Amiga, then eventually the Apple Mac and Windows PC.

So, what exactly did it do? Frankly, not a lot. By modern standards, early Cubase was hilariously basic. Until the early 90s, the software did little more than control external MIDI devices via a suitable interface. Updates focused on very basic usability functions like editing and rearranging MIDI phrases, mixing MIDI signals and sending SysEx messages.

Audio functionality followed for the fast new Power Macs in 1993, using hardware developed by Digidesign for their new Pro Tools system. Plugin support arrived in 1996, establishing the basic template we now think of as the modern DAW: MIDI sequencing, audio recording, audio effects and (eventually, with the implementation of VSTi in 1999) virtual instruments. Cubase development continues to this day, with v10 released last November and the program celebrating its 30th birthday this year.

Cubase is arguably the main player in the story of computer music as we know it. The modern DAW might have evolved in some way without the intervention of Rürup and Steinberg, but there's no doubt they were instrumental in defining the state of 21st century music technology.

Highly logical: Emagic Logic

The story of Cubase runs parallel with that of its main rival, Logic. Both trace their roots to the '80s, both originated from small developers in Germany, and both are still made today.

Like Cubase, Logic evolved from a simple mid-80s C64 MIDI sequencer, C-Lab Softtrack 16+, released in 1985. It went through a few iterations before it became what we now know as Logic, with Softtrack first spawning C-Lab Creator and Notator for the Atari ST, before lead developers Gerhard Lengeling and Chris Adam left to form Emagic and release Notator Logic in 1993, eventually renamed Logic.

Despite their similarities, there were also big differences between the proto-versions of Logic and Cubase along the way. Whereas Cubase was initially quite linear and track-based, Creator and Notator took a more pattern-based approach, allowing users to string together blocks of MIDI data to create song arrangements. The approaches aligned after the release of Notator Logic, and as both Cubase and Logic matured with the addition of similar audio functionality, plugin effects and virtual instrument capabilities.

Anyone who started making music after 2002 will know that Logic is only available for Mac OS, but this wasn't always the case. Early versions of Notator Logic were on the ST platform but shifted over to Mac and Windows as the Atari gradually became obsolete.

On July 1 2002, Apple announced their acquisition of Logic's parent company Emagic, lock, stock and barrel. Lengeling and team moved to California to oversee the Logic project (and related work). Apple's software is only ever available for Mac OS, so Windows development shut down immediately and Logic for Windows died with version 5. This is the crucial moment in the Logic story, defining Apple's interest in DAWs and shifting development exclusively to Mac OS.

What's notable about Logic is its price, relative to equivalent DAWs. Logic Pro X retails at £199.99. Compared to Cubase Pro 10 at £480 and Ableton Live Suite at £539, you can see how much Apple subsidise their software. So, not only is Logic a highly developed package, but it's a bargain for Mac users. From humble C64 roots, it's now one of the modern DAW giants.

From live looping to studio staple: Ableton Live

Necessity is the mother of invention, and that's certainly long been the case in music. Whether it's guitar players developing electric pickups and amplifiers to compete with the increasing volumes of big bands, or Kraftwerk developing their own drum machines and synths, countless technological innovations have been created by frustrated musicians. The same can be said of Ableton Live, initially developed by Bernd Roggendorf with Gerhard Behles and Robert Henke, also known as techno duo Monolake.

Behles and Henke had previously designed a step sequencer, PX18, for their own use. Coded using Cycling 74's Max visual programming language around 1995, PX18 was in some ways a prototype version - or at least a forerunner - of Live, with a vaguely familiar user interface. You can still find it on the Cycling 74 website, in fact.

Within a couple of years, the Monolake chaps had moved on to develop a different piece of software with a focus on sample playback. Early versions of Live were basic affairs by modern standards, but still a big step up from the simplicity of PX18, and clearly following their own path, with the focus very much on looping and triggering samples, timestretching and live manipulation of sequences and sets.

High performer

Live is an adolescent DAW compared to Logic, Cubase and co, the company being founded in 1999 and its first commercial release only arriving in 2001. As the name implies, its roots are in live performance as opposed to the studio sequencing and recording tasks that dictated the development processes of most other DAWs. The original motivation for developing the software stemmed from a frustration at the limitations of existing tools for performing with samples. Live's entire clip-based workflow is derived from that initial raison d'etre, offering what was, at the time, a completely new approach to live performance. It's still one of the most versatile, and effective ways to perform electronic music live.

Non-linear thinking

We now think of Live as a fully-fledged DAW in its own right, but a lot of its key features arrived much later than you may realise. MIDI sequencing and virtual instrument support weren't even added until version 4, which shows just how laser-focused the first few versions were on the idea of warping, triggering and arranging samples and loops.

If Live didn't completely tear up the rules on how a DAW should work, it turned a blind eye to many existing conventions. The software's real-time warping and sound processing options are extensive, but the Session View grid is possibly Live's greatest contribution to music making. It's a completely different approach to most DAWs, breaking away from the linear sequencing format and changing the way we think of arranging music (the complementary Arrangement View is closer to the traditional linear timeline used by most DAWs).

If you see an electronic act perform live these days, there's a very good chance that Ableton Live will be employed somewhere in their setup. But even more importantly, Live has become a studio mainstay for electronic producers and anyone who feels restrained by the strict linear timeline of so many other DAWs. Many users find it to be the most intuitive, user-friendly way to approach making loop-based electronic music, which is unsurprising considering its techno roots.

Back in the garage: how Logic spawned GarageBand

Apple's entry-level DAW is much younger than its big brother, but GarageBand shares more DNA with Logic than you might realise. When Apple acquired Emagic in 2002, the company's co-founder, Gerhard Lengeling, moved to California to take on a new role as senior director of software engineering (music applications).

Apple play their cards very close to their chest when it comes to software development, but it's widely believed that a big part of their reason for buying Emagic was to leverage the technology behind the Logic software itself, plus the expertise of Lengeling. Within a couple of years, Lengeling and his team had developed Logic 6, but also another entirely new DAW.

Announced by Steve Jobs in his keynote speech at the 2004 Macworld Expo, GarageBand was notable for the fact it was immediately bundled free with every Mac, meaning that all Mac users have a decent user-friendly DAW to play with. But don't assume that GarageBand is only an entry-level toy; under that friendly surface, it's powerful, and since its release it's been used to demo or even record countless commercial releases (see Rihanna's giant 2007 hit Umbrella, based on GarageBand's 'Vintage Funk Kit 03' drum loop).

It's also notable that GarageBand is Apple's choice for the iOS platform. You can't run Logic on your iPad or iPhone, but the mobile version of GarageBand is a surprisingly capable piece of software, with the ability to export projects to the desktop version. GarageBand marked a major shift in the landscape of music software, democratising and opening up the possibilities to a bigger audience than ever before.

Storm in a teacup

Of all the DAWs we lost along the way, the strangest of them might have been Arturia's Storm. Released in early 2000, Storm was kind of a DAW but also kind of not. In the end, that was probably its undoing.

Described by Arturia as a virtual studio, Storm combined sequencing, instruments and effects in a combined package, with Roland-inspired drum machines, acoustic samples, various vintage synth modules and loop players.

The first version focused heavily on combining pre-made loops and sequences to create tracks, but Storm evolved (slightly) into something capable of creating original tracks from scratch. Storm was best used when Rewired to another DAW, but if you're going to all that trouble, why not simply use a better DAW in the first place? Its closest rivals were other young, fresh DAWs like FruityLoops (now FL Studio) and Propellerhead Reason; unfortunately for Arturia, they both just did things better.

Storm made it as far as 2004's version 3 before Arturia finally cut their losses and pulled the plug to focus on virtual instruments. Storm remains a curiosity rather than a long-lost classic. Its best features made it into other Arturia plugins. Plugins you can use in a proper DAW.

The dawn of plugins

One basic requirement of a modern DAW is that it should act as a plugin host. Any half-decent DAW is expected to include a range of built-in virtual instruments and effects, and let you call up plugins from third-party developers. However, most sequencers and DAWs didn't have plugins, virtual instruments or even audio effects until well into their evolution.

The potential of digital signal processing (DSP) emerged during the 1960s, as computer technology allowed telecommunications companies and scientific researchers to start experimenting with audio signals. New York effect specialists Eventide were the first to bring this radically new technology to the studio with the release of their primitive DDL 1745 digital delay line, but it was their Harmonizer H910 multieffects unit, released in 1975, that really established digital technology as the future of studio processing. Over the next decade, digital effects became more common and affordable, while mass-market digital synths and drum machines also eventually emerged from the early 80s onwards (a few options such as the Synclavier and Fairlight existed in the 70s, but these were hugely expensive).

Despite the rise of digital audio hardware, the idea that this kind of DSP could take place natively within a music sequencing application running on a personal computer was unrealistic at best; early sequencers offered no audio capabilities whatsoever, while early version of Pro Tools relied on dedicated DSP hardware for their audio functionality. As hard as it might seem to understand from a 21st-century perspective, the ability to sequence and synchronise external instruments via MIDI was enough of a revolutionary idea in its own right. Digital audio processing in software followed eventually, but it took a couple of decades to mature and become ubiquitous.

VST's arrival

As far back as 1982, Eventide were the first to use the word 'plugin' in an audio context, although in their case it referred to physical EPROM chips that could literally be plugged into the SP2016 effects unit, allowing the user to add new programs to the hardware. The widespread proliferation of plugins as we now know them only really became viable once computer processing power caught up with the potential of digital signal processing in the mid 1990s.

The real turning point came in 1996, with the introduction of the standardised VST (Virtual Studio Technology) plugin spec for effects. Steinberg released the format to the world alongside Cubase 3 as a cross-platform standard, encouraging others to follow suit and establish an industry standard. Three years later, the VSTi virtual instrument specification was added. 13 years on from its initial release, VST and VSTi are still going strong, updated accordingly and flanked by Audio Units for Mac and the AAX format for Pro Tools. The last major DAW to hold out against third-party plugins was Propellerhead Reason, which only succumbed to VST support in 2017. "VST support in Reason? When hell freezes over," joked Propellerhead's then-CEO Ernst Nathorst-Böös as he announced its arrival.

Whatever your DAW, you can now take your pick from thousands of plugins, ranging from simple freeware to big-budget virtual instruments and pro-grade effects.

Sequencers vs DAWs

A lot of musical terms have been abused, misused and otherwise mangled over the years. If you want proof, just try explaining to a guitarist that the vibrato effect on their amp is actually tremolo, and that wobbling the tremolo arm on their guitar actually creates vibrato. Confused? We don't blame you...

Strictly speaking, the difference between a sequencer and a DAW is simple. The clues are in the names: a sequencer sequences or controls instruments using MIDI data or other control protocols, while a DAW handles digital audio.

In this case, though, we can forgive the fact that the two terms are sometimes used interchangeably, given that so many modern DAWs started life as sequencers. Some of the biggest DAWs on the market - notably Logic and Cubase - began as sequencers years before audio functionality was added. Others, like Pro Tools and Ableton Live, were always DAWs, but to some extent their ability to sequence instruments (virtual or hardware) is also pretty inherent to the concept.

Strictly speaking, they're all DAWs with sequencing capabilities, but if you refer to Logic or Live as a sequencer, people will probably still know what you mean. Now, what was the difference between vibrato and tremolo again?

Crackers and trackers

The 1980s were a strange period in music history. The music of the '60s and '70s suddenly seemed incredibly dated as synths, drum machines and samplers began to take over. Music videos emerged as a fresh, bold vision of artistic expression, while musicians began experimenting with sounds that had never been heard before.

The biggest problem was that for much of the '80s, computer technology couldn't keep up with the promise of a brave new future. Computers were just about starting to establish themselves as an option for sequencing and recording, but for some, it wasn't enough. One side of the story is self-explanatory and written into history as the simple process we've described elsewhere: sequencers let people program hardware, DAWs eventually evolved into what we now know as music software.

But there's another story that runs in parallel, lurking in the shadows away from the mainstream. Rather than professional studios and slick music videos, it revolves around early video games and the so-called 'demoscene', a subculture based on showing off programming skills via music and graphics, often going hand in hand with software cracking and piracy. Whether legit or not, programmers exploited the basic musical power of the Commodore 64 and the Amiga to showcase their talents, using incredibly clunky software such as Superions Future Composer and SIDMON.

Following the release of the (marginally) more intuitive Ultimate Soundtracker for Amiga, the whole software category became known as trackers, and the tracker scene evolved separately to the more user-friendly sequencer market. Typically based around numerical data entry and vertically scrolling timelines, trackers seem alien to most modern DAW users, but they've remained popular among a relatively small, hardcore group of users, with programs like OpenMPT and Renoise still available.

There's a crossover with the even more hardcore chiptune scene, in which producers push the limits of the sound chips of arcade games and consoles. Both have been responsible for big hits: Calvin Harris's first album was reportedly made entirely on OctaMED for the Amiga 500, while Timbaland settled out of court with Finnish demoscener Janne Suni after his production on Nelly Furtado's Do It was found to be notably similar to Suni's chiptune track Acidjazzed Evening.

What the tracker, demo and chiptune scenes might lack in size they make up for in moxie.

Acceptable in the '80s: Amiga and Atari

Life is pretty simple now. If you want a new computer, you probably ask your mates whether to go for a Mac or a Windows PC. For the small minority who want something leftfield and techy, Linux might be a contender.

In the 80s, however, things were different. Depending on your budget and what you intended to do with your cutting-edge personal computer, you could take your pick from Sinclair Spectrums, Apples, BBC Micros, IBM PCs, Acorns, Amstrads, Commodore 64s...

Most crucially, almost none of these platforms were compatible with one another. Hardware and software were extremely restrictive by modern standards, and in many cases, users were expected to demonstrate a basic understanding of programming and command-line operating systems just to carry out the most basic tasks.

With the shift from 8-bit to 16-bit computing, the pack was whittled down to a few key players. Notably for musicians, 1985 saw Commodore release the Amiga range, while Atari announced the ST. The two were fairly evenly matched in terms of processing power, but the Atari had one key advantage: a built-in MIDI interface, with 5-pin DIN input and output. Hook up a synth or a drum machine and you immediately had the makings of a basic home studio setup.

Unfortunately, the Amiga and Atari wouldn't ultimately survive for long. The eventual demise of both platforms had less to do with the capabilities of the machines than changing trends in the personal computer market.

Atari never fully recovered from the video game crash that took place in 1983, during which the console market became flooded and dominated by new players including Nintendo and Sega.

Commodore likewise failed to establish the Amiga as a serious business or professional option against the Macintosh and Windows alternatives, establishing a decent foothold in the formative digital graphics and multimedia industries before eventually filing for bankruptcy in 1994, clearing the way for Apple and Microsoft to dominate.

The importance of MIDI

Talk to anyone who tried to make music with synths and drum machines before MIDI and you'll hear tales of the horrors they endured.

Before MIDI, trying to control multiple pieces of hardware was, to put it politely, a pain in the neck. Synchronisation of clock signals was just about manageable, but sending CV (control voltage) signals to trigger notes was a different matter.

If you wanted to tune multiple pieces of kit or attempt anything polyphonic, things got even more tricky. It was more or less possible, sure, but you certainly needed more patience than most people could manage in a lifetime.

As for sequencers, they were basic. Monophonic analogue synths worked best with simple step sequencers, repeating the same basic 16th-note pattern over and over indefinitely. More advanced options existed, but would you really want them and could you even afford them? Roland's MC-8 Microcomposer, released in 1977, cost $4,795 - a little over $20k in today's money. It was reported to be hard to program and took the best part of an hour to back up a 3-4 minute sequence. Fewer than 200 units were sold.

The MIDI standard was released to the world in 1983 as a way for manufacturers to standardise the way their synths, sequencers and drum machines communicated with each other. (A quick reminder for those who haven't been paying attention for the last 36 years: MIDI is essentially a control protocol, allowing pieces of musical equipment to communicate with each other. There's no audio sent down MIDI cables, just simple instructions: play this note now; change that parameter; trigger this program change...)

Why is all of this relevant to DAWs? Because you can think of MIDI as the glue that held early sequencers together, allowing them to control other studio equipment. The same applies to modern DAWs: the nuts and bolts may be hidden in the background for the most part, but control signals based on the concept of MIDI remain the foundation of most 21st century music software.

Hardware sequencers still exist to this day in the form of gear like the Sequentix Cirklon or newly announced Pioneer Toraiz SQUID, but software has very quickly taken over as the easiest, most intuitive solution. In a curious twist of fate, MIDI has become relevant once again as trends have shifted back from 100% in-the-box software setups toward hybrids of software and hardware, but chances are you're going to want to control everything from your DAW. And you can. We have MIDI to thank for that.

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Early DAWs: the software that changed music production forever

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(Image credit: Fair dealing) Digital audio workstations - or DAWs - are now the mainstay of most producers' workflow, but despite their cutting-edge credentials, you might be surprised at just how far back their roots can be traced. In the early 1980s, the first, primitive versions of what we now call digital audio workstations began changing the way musicians could express themselves.

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Visiting Willem Twee Studios - a modern early electronic music studio

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[Sound on Sound]

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Headphone mixing is becoming a core skill for everyone. We asked some of the biggest names in engineering how they get the best results.

Every audio engineer I know loves the studio, and there's no denying that working in a 'proper' space, on studio monitors, is the most natural and comfortable way of doing our job. At the same time, though, we also need to be able to work effectively on the road - and to keep working during the Coronavirus pandemic, which has closed the doors of commercial studios worldwide. There are those who proclaim that mixing on headphones can't be done, but I know that to be false. So I reached out to some leading mixing and mastering engineers, to get their perspectives on the pros and cons of mixing on headphones, and their advice on how to make headphones a reliable means of monitoring when needed.

As freelancers, most mixers I know have spent their careers moving through various studios, and therefore have had to learn to be adaptable. "Engineers, by nature, we have to work under any circumstance and with whatever we have," says Manny Marroquin (Rihanna, Bruno Mars, Kanye West). "So I could adapt to mixing on headphones, if needed."

Much of the diverse career of David Bottrill (Peter Gabriel, Tool, Muse) has been spent in this nomadic lifestyle, leading him to believe that "You just tune yourself to your environment and the tools you have on hand. If a client has a deadline you just have to make it work."

This adaptability has cultivated a resourcefulness in most engineers that I've met, driven by the will to ultimately do whatever is needed to get a job done. Ali Staton (Seal, Madonna, Turin Brakes) exemplifies this in how he's managed to balance his life as a mix engineer with the work he does as Ableton programmer for artists like Madonna, Richard Ashcroft and Snow Patrol. "I did a record with George Ogilvie which started at Real World Studios, but then I had to be in the States with Snow Patrol and Ed Sheeran for nearly five months. That's when I started trying to mix on headphones, because I had to carry on working."

It clearly worked, because Ali ended up mixing part of the album back in his room at Real World Studios, and says: "You'd be hard pressed to tell the difference between the headphone and studio mixes, especially now that they're mastered."

Manny Marroquin: "I can't get dynamics as much as I do on my monitors. It's not to say I couldn't do it on the headphones, it's just harder and not my workflow.

These days, there's an abundance of great monitor speakers that are relatively neutral and flat. But in any studio, they're only as good as the control-room acoustics. Anything that comes between the sound itself and your ears will change your perception of the mix. This means that you are making decisions based on the biases of your monitoring and acoustics, which are manifest in alterations to frequencies, phase and imaging. Solutions such as Sonarworks Reference, IK Multimedia's ARC, Dirac and the offerings from Trinnov are amazing tools and are incredibly helpful for tuning a listening space but, ultimately, this variable in your perception is at the mercy of physics, which doesn't often work in your favour.

So, a mix is actually quite ephemeral and ever-changing, depending on where it's played back, and your job as a mixer is to create something that is the best representation of the material wherever it gets heard. This relies on your reference being as unbiased as possible, and the argument in favour of using headphones to mix is that it removes the most volatile of factors: room acoustics.

As the gatekeepers between the mixer and the final release, mastering engineers arguably have the best purview on this line of thought. "Assuming one has a good, calibrated room and monitoring system, I would never initially master or mix on headphones," says legendary mastering engineer Bob Ludwig (Led Zeppelin, David Bowie, Jimi Hendrix). "If one's speaker setup is not satisfactory, then one must rely on headphones, which can qualify as 'audiophile' quality for not crazy money." All of the mixers I spoke to share similar...

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Headphone mixing is becoming a core skill for everyone. We asked some of the biggest names in engineering how they get the best results. Every audio engineer I know loves the studio, and there's no denying that working in a 'proper' space, on studio monitors, is the most natural and comfortable way of doing our job.

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[Future Music]

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There are many differences between experiencing a track on speakers and on headphones* The best headphones for music making you can buy .
If you only ever use headphones for mixing, it's all too easy to misjudge your mixes. Common mistakes include making the stereo field too narrow, and not splashing enough reverb onto vocals and synth parts, leaving them overly dry. Judging how much of any effect to use should be a decision that's made through listening on both cans and monitors.

A well produced track should sound great on everything from laptop speakers to a high-end hi-fi. Don't forget: just because you can mix on headphones doesn't necessarily mean that you should.

That said, there are techniques that can help you mix more accurately when using just a sat of cans. In this walkthrough we'll guide you step-by-step through the process of mixing a track without using monitors.

Step 1: We're using some stems from a piano-led alternative pop track and have dropped them into our DAW. The track sounds pretty good in our headphones but we'll need to make some adjustments so that it both translates well over a monitor setup and has a bit more depth and detail for those listening on cans.

Step 2: To make the FX track stand out more in the mix, we use Brainworx BX Solo to widen the stereo field to 200% and drop the volume by -3dB. We use BX Solo again on the bassline but this time narrow the width to mono. We also add an instance of Waves MaxxBass to generate harmonics of the bassline, giving us an audible impression of the thump of the low end, even though we can't feel it.

Step 3: The arpeggio track will sound fine over monitors but is a little too sterile in our headphones. We give it some subtle ping-pong delay with KR-Delay CM's Delay set to 1/16 and the Dry/wet at 15%. Although this effect won't be noticed much when heard on speakers, it helps to make the whole track sound less dry for those listening on 'phones.

Step 4: The chords and notes of the piano line are easy to pick out using our cans. However, we'll need to make it sound more in-your-face over speakers. Instead of turning it up, we give it +8dB Gain using console channel emulator Satson CM. We drop the track's level to -8dB to compensate for this. We also increase Satson CM's High pass to 200Hz to stop the piano clashing with the bass and kick.

Step 5: For the same reason that we added a little depth to the arpeggio line with some delay, we're going to give the whole mix some reverb. This will help things gel together over headphones and give some extra depth. We use Reverberate CM's Close Drum Room preset and turn the IR Gain down to -30dB so that we get just some subtle reverb feedback.

Step 6: It's hard to tell if our track is EQ'ed well using just headphones, so we're going to match the EQ from a commercial track. Houdini by Foster The People has a similar kind of vibe, so we're using that as a reference. Using Ozone 5's Equalizer, we take ten-second snapshots of our own track and Houdini and hit the Match button. We adjust the Amount and Smooth sliders as pictured.

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There are many differences between experiencing a track on speakers and on headphones * The best headphones for music making you can buy . If you only ever use headphones for mixing, it's all too easy to misjudge your mixes.

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The '80s were rich with developments in the synthesizer and drum machine world that changed the way music was produced and heard. The instruments created by the Roland Corporation of Japan - what is now known as the x0x series - are integral to the legacy of the records they helped create. In many instances, the units were intended to be an accompanist for rehearsing or composing; it is safe to say that the designers did not imagine the revolutions in music they would help fuel. The drums did not sound like an acoustic drum kit, which led to many machines being discarded and resold in pawn shops - they were picked up by DJs and producers who would use them to invent new genres. Since most of these machines were produced in limited quantities of around 10,000, their secondhand market prices continue to rise, leading both hardware and software manufacturers to attempt to clone, imitate, and even expand on the original blueprints. Every one of these boxes is a classic in its own right, something that no set of samples can ever replace.

SH-101 [1982]

The plastic-encased, battery-powered, 32-key SH-101 is a monophonic analog synth that excels at basses and leads. With a very similar synthesis design to the MC-202, it has a single VCO with sliders that control levels for its saw, square, and sub oscillator waveforms. The square wave has an adjustable pulse width that can also be modulated by the LFO and the sub osc is selectable either one or two octaves below the fundamental. Its low-pass filter is capable of some wild self-oscillation and can be modulated from by the LFO or ADSR. One place where the 101 differs from the 202 is the inclusion of noise as a sound source. Another is the SH-101's LFO having four selectable waveforms versus the 202's sine-only design and a pitch bender/mod wheel that can be used for expressive modulations while playing the keyboard. The 101 has a simple, syncable sequencer up to 100 steps and a basic up, down, or up/down arpeggiator. On the back panel are 1/8" connectors for CV/Gate in/out and hold. Available in grey, red, or blue, the SH-101 could be worn with a strap and also used with a handle like a keytar to evoke the early '80s from whence it came. There have been a few software clones of the 101 and most recently, Roland created a "plug out" version of their own that works with the new AIRA series System 1 keyboard. It is also apparent that the Arturia Brute series synthesizers draw some influence from the 101 and 202. Juan McLean calls it his favorite synth and claims that he and James Murphy have three or more 101s in their live rigs. Long before that, A Guy Called Gerald used the SH-101, triggered by the TR-808, to create the acid house classic " Voodoo Ray " in 1988.

MC-202 MicroComposer [1983]

Essentially an SH-101 in a more compact enclosure, the MC-202 MicroComposer replaces full-size keys with two octaves of rubber buttons and upgrades the sequencer considerably. This is not to say that sequencing is very easy on the 202; when compared to any number of groove boxes that came later from Roland or other manufacturers, this unit can be a bit frustrating to say the least. The trade-in is the analog synth voice that it sequences and the 1-in/2-out DIN-sync connectivity to other Roland boxes. In addition to CV/Gate control on the back, tape in/out provides a way to save and recall sequences. This was cleverly used by the people at Defective Records to create MC-202 Hack, a software that allows for a Mac or PC to be used to create sequences. Probably because there is no mod wheel, the 202 has an LFO delay that the 101 does not, enabling the modulation to kick in after a note has been held for a while. The MC-202 has its similarities to the TB-303, and on the secondhand market it costs about half, making it a more reasonable option for many. For the more adventurous, many modifications are possible to the MC-202, essentially making it modular, with CV jacks for many of the synth controls and inputs/outputs at many points in the signal path. Notable users Autechre make frequent mention of the MC-202 as part of their setup.

TB-303 Bass Line [1982]

Not much can be said about the 303 that hasn't been said already. The little silver box was, almost comically, designed to sound like an electric bass player, and was initially a failure from a business perspective. Instead it launched entire movements, sub-genres, and styles with its 3-pole filter and single oscillator design - the sound of acid. Chicago's Phuture are often considered the first to create an acid house song with the seminal " Acid Tracks," where they tweaked the filter beyond the manufacturer's probable intentions to bring out the squelch that has been heard countless times since. With batteries, the memory can store 64 patterns that can be chained to form seven songs, but one bar loops are most common. Also one of the most definitive features is the ability to have accent and slide turned on or off per step. These sequenceable pitch bends and accents are largely responsible for the sound of this box. The 303 is often processed with a bit of distortion for an extra taste of harmonics. There are many modifications that expand the somewhat limited synthesis capabilities of the original unit - the Devilfish and Borg mods are some notable choices. Because of its scarcity and high demand, there are probably more clones of the 303, both hardware and software, than any other synth. Notably the new Cyclone Analogue TT-303 looks like an exact replica. Also popular are the x0xb0x and the Future Retro Revolution. Roland recently released the comparatively inexpensive yet digital TB-3, another member of the AIRA series, with a pressure-sensitive touch pad and sounds that mimic the original while also going way beyond it.

TR-606 Drumatix [1982]

Designed side by side with the 303 was its rhythmic companion, the TR-606. It's an analog beatbox with seven sounds and a single 1/4" output. There's individual level knobs for kick, snare, cymbal, and accent, while two toms and the open and closed hi hats utilize two level controls. The 606 kit has a similar sound to the TR-808 with less control over sound parameters. The sequencer is the 16-step format made famous by the Roland units and the memory can store 32 patterns with the ability to chain up to four of them together. There's also room for eight songs in its memory. Unlike its Roland drum machine brethren, it's able to jump between patterns while in write mode, useful for improvising live without pressing stop. The 606 has one DIN sync in and out for connecting to other Roland boxes. Acidlab, maker of several Roland x0x series clones, might be the first to release a TR-606 clone before 2014 is over. A cheaper alternative is the Boss DR-110, which sounds strikingly similar, although sync is only available with a modification. Many mods are possible for the TR-606 that include adding MIDI, individual outputs, inputs, additional trig outs, and timbral changes galore. A listen to Plastikman's first album Sheet One is a surefire way to hear a whole lot of 606. Aphex Twin, Squarepusher, Massive Attack, and a host of others also made heavy use of this machine.

TR-626 Rhythm Composer [1987]

Possibly the least coveted and least known of the lot is the TR-626, which uses 12-bit digital samples of 30 different acoustic drum kit and Latin percussion sounds. Although not as heralded as its predecessors' kits, apply a little processing and there's definitely something useful for everyone here. The 626 has stereo main outs (but no panning) plus eight individual outputs. Sounds are tunable seven semitones in either direction and volume adjustable via buttons, rather than knobs or sliders. The sequenced rim shot sends a pulse out of the 1/8" trig output, which can be useful for sequencing synths that have CV. The sequencer is very similar to the TR-707 and the 727, with the non-backlit LCD display utilizing dots to show where notes occur. Three different levels of accent are possible for each sequencer step. MIDI is available on the unit for triggering sounds, plus a beat clock. There is a rather sizable portion of the front panel, which circuit benders will find useful for adding a variety of switches, jacks, and pots. As demonstrated by the guys at Burnkit2600, easy access to the 32-point sample chip allows the samples to be mangled in all sorts of ways. Even more affordable, and also popular with the modders, is the TR-505 (released in 1986) which has only a stereo output and 16 non-tunable sounds. You can't exactly spot the 626 in a lot of famous songs, but Front 242 used the machine, specifically on 1988's "Circling Overland."

TR-707 & 727 Rhythm Composers [1984]

These two units are basically the same in their design and sequencing, but they have different sample banks: the TR-707 has the usual acoustic drum kit components plus cowbell and tambourine while the TR-727 has all Latin percussion sounds (bongos, congas, timbales, etc.). Each unit has 15 sounds in its memory. The orange and tan 707 is sometimes considered the "poor man's 909," since its hi-hats, clap and cymbals are very similar. The only sound parameter available on the 707 is volume, adjustable via small sliders on the top panel, and all its sounds are digital samples (8-bit mostly, with 6-bit cymbals). The 909 is both analog and digital. Both units use the LCD matrix sequencer screen for a clear representation of the programmed pattern. Sixty-four patterns and four songs can be stored in the memory which is doubly expandable with the M-64C cartridge. These machines can send/receive MIDI notes and beat clock and also have DIN sync in & out. Similar to the 626, there's an 1/8" trig out on the 707 that the rim shot sequences. The back panel has a host of 1/4" jacks with ten individual outputs, stereo outs, a headphone jack, and a start/stop trigger. Legend has it that Larry Heard used the trigger output to sequence the Juno 60 on his Mr. Fingers classics for Trax Records, while INXS used the 707 kit on the mainstream hit "Need You Tonight."

TR-808 [1980]

The granddaddy of them all was also the first to use 16-step sequencing, containing the bass drum that put the "boom" in hip hop and electro from the very beginning. The 808 uses analog synthesis to generate its tones, which may be the key to its huge sound - its kit is quite possibly used on more hit records than any other drum machine. The signature bass drum has controls for tone; actually, a CV trigger that creates an attack sound and a decay knob that extends the trunk-shaking kick. The crispy snare, hand clap, hats, cymbals, and cowbell are all instantly recognizable. Tunable toms and congas make melodies within the percussion tracks. All of the 12 voices have volume controls and individual outputs, and there's three trig outs. Up to 64 patterns and 12 songs can be stored, with 16 of those patterns used as drum fills. A single DIN sync connection can be switchable to master or slave but it is not difficult to install a MIDI retrofit. First appearing on Yellow Magic Orchestra's live version of "1,000 Knives," the 808 really crept into the world's consciousness in 1982 when Arthur Baker and Afrika Bambaataa reprogrammed the beat from Kraftwerk's "Numbers" to make the b-boy anthem "Planet Rock." Marvin Gaye's Sexual Healing, also released in '82, was a mainstream album that utilized the 808 for all of its drum sounds. Juan Atkins also used it in his early years (as Cybotron), as did his Detroit contemporaries. Egyptian Lover made it his staple drum sound on the West Coast, and the Beastie Boys, LL Cool J, and nearly all Miami bass producers made sure it dominated the East. Over the years there have been many hardware and software clones with varying authenticity, but there's nothing quite like the real thing.

TR-909 Rhythm Composer [1983]

Although the Roland machines can be heard on records of all genres, the 909 kit is mostly associated with the history of techno and house. It has a hit-you-in-the-chest kick drum that is more punchy than booming, as well as an open hi hat with adjustable decay that sounds perfect on the upbeat every time. The bass drum, snare, three tunable toms, claps, and rimshot are all analog-produced, while the hats, ride, and crash are 6-bit samples. Like the 808, each of the 12 drum voices have volume pots and individual outputs that remove them from the main output when a cable is inserted. Most of the instruments have other sound parameters too; only the rim shot and claps do not. The 909's seven different levels of swing contributed to the funky feel of countless house records - while this feature was not available on the 808 or 606, it was later implemented in the 707 and 727. Mods for tuning the hi hats and extending the range of the bass drum exist and aren't difficult to pull off, if you're comfortable with messing with what is now a very expensive machine. The 16 pattern/instrument keys are like old-school computer keyboard keys with an LED in the top center and are the most playable of the x0x series. Up to 96 patterns and eight songs are storable in the memory and that is expandable with the 64C cartridge. One key feature of the 909 is that it has DIN sync in, MIDI in, and two MIDI out ports, making it connectable to several machines. Mantronix and Schoolly-D are notable hip hop artists to use the 909 kit while pretty much all techno producers (especially from Detroit and Chicago) have utilized is classic sound. Jeff Mills has been known to destroy a dancefloor with a 909 between the decks.

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The Red Bull Music Academy Daily is the online publication by the Red Bull Music Academy, a global music institution committed to fostering creativity in music. Just like the Academy, we think of it as a platform for the essential ideas, sounds and people that have driven - and continue to drive - our culture forward.

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Learn all about Logic's new meter.

With the release of 10.2.3 Logic released the Loudness Meter, this has been much needed for a long time and is something I personally am very happy to see added to Logic's huge range of amazing plugins.

Although it's a simple to use plugin there's a lot to understand and learn about why it's been added and what it means for you and your music. It will help you to judge how loud you music will be heard by your listeners and how loud your music is compared to other tracks in your style.

History of Loudness in brief - really brief!

I'm sure you'll wanna get straight to the video so here's a very quick why this plugin came into being.

Logic's Loudness Meter is really about adding a plugin that will be widely and mainly used by broadcast professionals. Ever watched your favourite show and when the adverts come on they suddenly seem much louder? It's all to do with limiting and crushing the dynamic range of the music. Because of the digital limiters ability to get ever louder and most commercial meters inability to read the true loudness of the music a new meter was needed that could read long term average levels of a piece of music so that all broadcast material would be heard by the viewer at the same perceived audio level.

Got it?

The loudness meter was introduce to combat that specific problem.

In this first part looking at the Loudness Meter I go into more detail about the plugin and some of it's key components.

Check out the video and enjoy!

Lou dness Meter All About Broadcast

Rather read about the plugin than watch the video? No problem read on 🙂

PPM, VUdB, dBFS, dBTP with so many ways to measure the strength of audio why do we need another, LUFS? Isn't it all just a bit too confusing and is it relevant to you?

Older forms of metering still have a firm relevance in music recording and production, broadcast media though has an Achilles heel and that's the standardisation of audio levels across content.

Advertisers, like record labels, loved the loudness war because if their advert could be 2 or 3dBs louder than the advert before which do you think would likely stand out more?

With adverts getting ever louder than your favourite TV shows people spent more time grabbing the remote as levels constantly changed between content. Pretty annoying. So the brains at something called the International Telecommunication Union or ITU scratched their heads and came up with a way to standardise audio across content.

Hopefully I've not lost you or put you off just yet because trust me there's some cool stuff coming up!
LUFS - Your Ear Is In Logic's New Tool!
Before I go any further let's briefly talk about how Logic's Loudness Meter reads levels.

As discussed in previous chapters there are two basic ways to measure audio using the Level Meter, RMS and Peak.

RMS gives you an indication of the average level and Peak the loudest.
Plugin Overview The In terface
So here's the rub, RMS tells you what the average is but can't read long term average values from start to finish; Peak gives you no indication of how loud your music actually is you can only gauge loudness if it's heard for a relatively long period of time.

Our ears are essentially wired to be long term averaging meters and that's exactly what the Logic Loudness Meter aims to do. To measure audio signals like your brain does when sound passes through your ears.

The Loudness Meter has three ways to measure amplitude. Momentary, Short Term and Integrated.

Momentary - Has the fastest response of around 400ms and essentially allows you to monitor the peaks of transients like snares or guns shots for example if you're dubbing a film.

Short Term - Has a longer response time measuring average levels over several seconds similar to RMS giving an indication of the short term average level.

In tergrated - For Integrated think of your music from start to finish, view it like a landscape with the music's average level rising up and dropping down over time from beginning to end.

Integrated measures the average level of the entire music providing a more accurate reading similar to the way we perceive loudness as humans!

Integrated only works once you click the start button and play your track from beginning to end. The plugin will analyse the entire mix and give you an Integrated value and LU range (LOUDNESS UNIT RANGE). Values and readings can be reset by clicking the reset button. We'll look at what all this means shortly.

In order of importance the Integrated value is king followed by Short Term and then Momentary but of course the 3 work in tandem.

In the next video let's look at three different tracks from Logic's sound library and compare their loudness levels using the Logic's Loudness Meter.

You can get part three by clicking Logic's Loudness Meter Part 2 and 3.

About miL admin

I am a composer, songwriter and mixing engineer, my music has been broadcast and released all over the world; I have written for TV, advertising and publishers globally including Bdi Music, Disney, Discovery Channel, BBC and many more.

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With the release of 10.2.3 Logic released the Loudness Meter, this has been much needed for a long time and is something I personally am very happy to see added to Logic's huge range of amazing plugins.

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120 Years of Electronic Music* is a project that outlines and analyses the history and development of electronic musical instruments from around 1880 onwards. This project defines 'Electronic Musical Instrument' as an instruments that generate sounds from a purely electronic source rather than electro-mechanically or electro-acoustically (However the boundaries of this definition do become blurred with, say, Tone Wheel Generators and tape manipulation of the Musique Concrète era).

Donate!

'120 Years of Electronic Music' has always been and hopefully always will be a free and open resource. The site has been hosted for free for many years by SPC London ; Please consider a donation towards the hosting costs to keep this project free for everyone:

Thank You!
Simon Crab - 120 Years of Electronic Music

The focus of this project is in exploring the main themes of electronic instrument design and development previous to 1970 (and therefore isn't intended as an exhaustive list of recent commercial synthesisers or software packages.) As well as creating a free, encyclopaedic, pedagogical resource on the History of Electronic Music (and an interesting list for Synthesiser Geeks) my main interest is to expose and explore musical, cultural and political narratives within the historical structure and to analyse the successes and failures of the electronic music 'project', for example;

Modes of interaction for performers and composers: Atonality and just intonation as a theme in instrument design.

The ideas put forward in Ferruccio Busoni's 'Sketch of a New Aesthetic of Music' (1907) inspired a generation of composers to explore micro-tonal and varied intonation and Hermann von Helmholtz's 'On the Sensations of Tone' (1863) provided an understanding of the physics of sound suggesting the possibility of creating an unlimited palette of tones and shapes beyond the restriction of traditional instrumentation. This lead directly to the design of several new instrument; Thadeus Cahill's Telharmonium (1897) and Jörg Mager's Sphäraphon (1920s) amongst many other, that explored new forms of interaction freeing the composer and musician from the 'tyranny' of the fixed tempered Piano keyboard (which at the beginning of electronic music instrument design was a fairly recent standard). Though this experiment was ultimately doomed due to commercial pressure on instrument designers to provide simulations of existing instruments on a fixed tempered scale for popular music, the concept survived into the 1960s in 'serious' experimental music with the era of the Electronic Music Studio; GRM, Milan, WDR, Columbia-Princeton Electronic Music Center etc. and even Moog (in the original instruments) and Buchla's modular synthesisers. More recently interest in atonality and non-manual control has re-surfaced with software synthesis and audio computer languages.

New composition tools: Musician-free composition

As well as developing new physical interface models another channel of exploration was to use electronic musical instruments to free the composer from problems associated with performers and performance; removing the musician from the process of musical production allowing the composer to create pure unadulterated music concepts. When this was first proposed - first proposed in Russia in the 1930s by Avraamov and later Hanert's ' Electric Orchestra' (1945), Percy Grainger's ' Free Music Machine' (1948) and Murzin's ANS Synthesiser ) this concept was considered extreme, utopian and unachievable and met considerably hostility from traditionalists This goal has mostly been achieved through digital software but what has been the effect on composition?

An exploration of the dichotomy between radical culture and radical social change: the role of the avant-garde in political movements

After the 1917 Bolshevik revolution in Russia, a utopian avant-garde movement inspired by Futurism and Anarchistic ideas developed which included radical new music and performance. New, atonal, music was written, new types of theatre evolved and new instruments invented to bring about this utopian age; Arseney Avraamov conducted a huge symphony of sirens involving warships, factory hooters and artillery and proposed the destruction of all pianos to free music of fixed tonation. It was proposed that a new scientific culture replace the old archaic cultural order- brought about by new electronic instruments and scientific investigations into audio and visual perception (this period is covered in some detail in Andrei Smirnov's excellent book ' Sounds In Z '). The Bolshevik government, traditionalists at heart and worried at their lack of control of this anarchic movement, suppressed it replacing it with manageable propaganda based culture of uplifting, popular socialist realism; many of the former avant-garde were murdered, imprisoned or sidelined for the rest of their lives.

Similarly, in the early period of Nazi Germany (1933 onwards), Hitler's government made moves to embrace Futurism and Modernism (for example; Marinetti's Berlin Futurist Exhibition of 1934, Sponsored by Goebbels and the Nazi Kdf organisation and the final Berlin Bauhaus before it's exile to the USA). Modernism was a key component of Fascism and early National Socialism; technology was promoted as a National Socialist principle; the Volkswagen ('Peoples Car'), Autobahn's, Film, Mass-entertainment (Controlled through the Kdf 'Strength through joy' organisation) and electronic music; the first every electronic music orchestra ' Das Orchester der Zukunft' at the Nazi-sponsored Berlin IFA in 1933. Again, this avant-garde movement was finally suppressed in favour of conservative fascist popular culture during the late 1930s.

'120 Years Of Electronic Music' is an ongoing web project initiated in 1995 by the author and musician simon.crab@gmail.com . The project is completely non-commercial and self-funded. The aim is to make the information available on a free/open source basis i.e. you can use content in any way as long as the content owners are credited and it would be courteous if you could acknowledge the source by linking or referring to the site.

Simon Crab, Hastings, UK. 2019

*The name: '120 Years of Electronic Music' project was begun in 1996; considering electronic music started around 1880 this was quite an accurate title for the time. Almost twenty years later it's a bit out of date but it's become something a bit of a brand-name and hard to change

Update:

A Chinese language version of '120 Years of Electronic Music' (by Guangyu Dong at the Communication University of China School of Journalism (SSI), Beijing.) is now available here.

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120 Years of Electronic Music* is a project that outlines and analyses the history and development of electronic musical instruments from around 1880 onwards. This project defines 'Electronic Musical Instrument' as an instruments that generate sounds from a purely electronic source rather than electro-mechanically or electro-acoustically (However the boundaries of this definition do become blurred with, say, Tone Wheel Generators and tape manipulation of the Musique Concrète era).

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To celebrate the magazine's 30th birthday, the SOS team take a trip back in time, finding out what it was really like to work in a mid-'80s home studio.

Nostalgia is a kind of selective memory. From our 21st-century perspective, the past stretches out like a golden landscape of great music, classic gear and bizarre haircuts, and we tend to overlook the fact that there were plenty of dreadful records in the '70s and '80s - and that many of the great ones were made despite, not because of, the equipment then available. It's also easy to forget that the gap between home and professional studio environments was once a gaping chasm. Much of the equipment that is revered as 'vintage' today would have been available only in professional studios in 1985, and far too expensive for most home-studio operators.

With that in mind, we thought that a fitting way to mark SOS's 30th anniversary would be to revisit home recording as it really was in the mid-'80s. We set out to assemble a setup that would include only equipment that was realistically within the reach of the average SOS reader, and use it to program, record and mix a complete pop song - in an '80s musical style, naturally!

Sound On Sound was launched in response to a revolution in project-studio music recording. The advent of the MIDI protocol, MIDI sequencers, timecode-based synchronisers and narrow-gauge multitrack, all in the first half of the 1980s, meant it was becoming practical for hobbyist musicians to create complete tracks. Suddenly, eight- and 16-track recording was no longer solely the province of professional studios; but perhaps just as significantly, sequenced MIDI synth sounds and drums could both augment track count and maintain audio quality by being run into the final mix as synchronised live sources, rather than being recorded to multitrack.

Instruments such as the Yamaha DX7, Roland Juno 106 and Korg Polysix were bringing polyphonic synthesis to a mass market; Akai and Emu were doing the same with sampling, and with a little imagination it was possible to extract plausible rhythm tracks from a new generation of drum machines. There was a good selection of relatively affordable mixers to tie everything together, and even luxuries such as digital reverb were starting to become a realistic prospect for home-studio operators. For the first time ever, talented amateurs could create sophisticated recordings without the help of other musicians or engineers.

Yet, at the same time, many other technologies we now take for granted still lay in the future. Affordable sample-based synths and workstation keyboards would not appear until the later part of the decade. Steinberg had yet to create the ubiquitous Arrange Page, the graphical template to which today's sequencing software adheres. Digital recording was in its infancy, so not only were luxuries such as non-linear editing and plug-in processing unknown, but unless you were lucky enough to own a Sony PCM F1 converter, master mixes had to be recorded to quarter-inch analogue or cassette tape. And entire market sectors that thrive today, such as active monitors, acoustic treatment and affordable capacitor microphones, had yet to be created at the launch of SOS.

Of those SOS staff based in our offices near Cambridge (a list that does not include founders Ian and Paul Gilby, Editor in Chief Paul White or Technical Editor Hugh Robjohns), only Editorial Director Dave Lockwood worked in '80s home studios first time around. His recollection of the typical home-studio setups of the time - described in the 'What Would You Find In An '80s Home Studio?' box - became our template in setting up the studio. His attic also proved an invaluable source of period-correct equipment, including the centrepiece of the entire setup: an Atari 1040ST home computer, complete with MIDI synchroniser and a copy of C-Lab's Notator sequencer.

Other team members were able to beg, steal or borrow a few other typical home-studio items: a 16-channel Soundcraft 200SR mixer, a pair of Yamaha REV7 digital multi-effects units and a Fostex M80 quarter-inch eight-track tape machine. However, although we could muster various 'vintage' synths between us, what none of us had was a suitable selection of early MIDI instruments. With that in mind, we turned to leading UK hire firm FX Rentals, who very kindly lent us several key pieces from their vast treasure trove of music gear: an original Yamaha DX7, a Roland Juno 106 polysynth and TR707 drum machine, and an Akai S950 rackmounting sampler.

Assembling this lot brought home how unwieldy the home studios of yore were. Our setup might have been a typical 'bedroom studio', but anyone who managed to fit that lot into his or her bedroom would struggle to do any actual going to bed too! Even in SOS's large meeting room, it was a challenge to arrange it all in such a way that all of our cables reached, and everything was accessible, while any changes to the setup would plunge us into feverish sessions of patching and re-patching. Since the Soundcraft mixer was a live-sound variant with no tape returns and only four groups, this had to be done on a regular basis. In some cases it was also a headache to find appropriate cables: the main outputs of the mixer, for example, appeared only on male XLRs, which needed somehow to be connected to RCA phono inputs on our master recorders and be paralleled to our monitors.

Our first challenge was to find a suitable song to work on. We toyed with the idea of covering an actual '80s hit, but this would have posed copyright problems, and we didn't want to be tied to slavishly recreating existing sounds. Nor did we want to begin with a song that had already been demoed or arranged using modern equipment and techniques. We also felt it would be nice to have a lyrical theme that tipped its hat to the decade in question. Reviews Editor Chris Korff supplied that with his genius title 'Love Is A Two Player Game', and from there it was a relatively straightforward process to weld some vaguely suggestive '80s video-game references to cheesy extended chords of the kind favoured in '80s white soul. I made a deliberately rough demo with vocal and acoustic guitar, leaving lots of space that could be filled in later with characteristic '80s arrangement ideas.

As we started the project before our hire synths arrived, Dave Lockwood and I decided to try to get a head start by hooking the Atari up to a Yamaha MU50 General MIDI sound module, and roughing out the MIDI arrangement.

"The software fired up first time. Amazing. It wouldn't always do that when it was new. Then I installed my E-Magic Unitor SMPTE sync box, causing Notator to hang at the end of loading. Cleaning the tarnished connectors worked its usual magic and we were up and running. Now all I had to do was remember how to work it all.

"To be strictly 'period correct' it should probably have been a less sophisticated SMPTE sync box - something like an XRI Systems unit, maybe - and Steinberg's pioneering Pro 24 doing the sequencing, but we had the Unitor and knew that it worked. Creator/Notator was also what I spent many years using in the mid-to-late '80s, so even though I hadn't looked at it for the best part of 25 years, I reckoned it would all come back pretty quickly.

"Notator (Creator is the same sequencer without musical notation) is a pattern-based sequencer with a very powerful Arrange mode for song construction. Becoming reacquainted with it for this project reminded me just what a powerful tool that mode is when building the early shell of a MIDI arrangement. No modern DAW is quite so fast at changing the length of whole sections or adding repeats with small variations. Huge structural changes can be made with a single numeric variable.

"Tempo was established using the system click, and then a simple shell drum part, step-programmed with kick, snare and hi-hat. In recognition of the number of sequencer-based '80s hits that were between 120 and 125 bpm, 4/4 time, with drums and keyboards hard-quantised to 16ths, we created a song based on exactly that, settling on a tempo of 123bpm.

"Looking at the 16 empty tracks of Pattern 1, I could remember that step writing was done in the track Edit window and, eventually, that notes were inserted by dragging the Note icon (pleasingly, actually the word 'Note') to the work area where they appear with the usual parameter set: note pitch, MIDI channel, velocity and position."

"The vintage keyboards and drum machine that we were going to be using had not arrived in time for our programming session, so we were temporarily using a hardware General MIDI module for sounds. Nonetheless, we were careful to choose sounds that were as close as possible to the ones we would actually be using. Our GM module had a DX-style Rhodes and an analogue synth-like pad that would cover the work of the DX7 and Juno 106 that we were awaiting. The note allocations for the drums, which were to come from a Roland TR707, were reliably standard, although I miscalculated both the number of toms and crash cymbals, and forgot that the actual 707 ride cymbal would sound like a trash can being hit with... probably another trash can. We would rethink that one in the later programming.

"At this stage, we still planned to use real bass guitar, so a very basic synth bass playing eighth notes gave us our basic feel for the programming work.

"With Arrange mode turned on, patterns are placed in a list that is played out sequentially, so our basic working method was to create a pattern for each of our major sections: intro, verse, chorus, and so on. The simplified harmonic shape of the piece was outlined with the pad sound, and the Rhodes-like voice used to replicate the distinctive DADGAD-tuned acoustic guitar part Sam had used to write the song. Some of those voicings were then simplified in deference to parts that we thought would be added later.

"Notator's Arrange mode has a set of track mutes that are associated only with Arrange entries, not the patterns themselves. This means you can use the same pattern a number of times whilst muting and unmuting different tracks each time it is played - perfect for a simple song structure where you might want to vary just the drum fill into the next part. Nine patterns covered all the sections, given the variations that could be generated by the Arrange mutes, and in an afternoon we were able to create a complete framework for the song that could be used to program more detailed MIDI parts and ultimately sync to tape for audio overdubs."

Even on the much-maligned General MIDI sound palette, our draft arrangement sounded pretty good, with appropriately brash drums, lush keyboard pads and punchy synth bass. We couldn't wait to hear the effect of translating it onto period-correct, sought-after vintage instruments. With a sense of real anticipation, we unpacked the DX7, Juno and TR707 from their flight cases, dug out yet more MIDI cables, selected the classic Rhodes piano and string pad patches and pressed Play.

It was awful.

The richness and density of our arrangement was completely lost. Classic synths though they are, the DX and Juno sounded thin and weedy in isolation. Compared with the MU50, the DX seemed to have an exaggerated response to note velocity, which made even a simple keyboard part sound horribly uneven, while the Juno simply wouldn't play even a basic melody part without randomly dropping notes. And although the TR707's chunky kick and snare sounds held up well, its MIDI spec was about as basic as it could be, more or less responding only to Note On messages and nothing else.

Although we did have access to an original Korg MS20, we didn't have a MIDI-CV converter handy, so we cheated slightly and pressed the reissue MS20M into service on bass. As if to show solidarity with the instruments of 30 years ago, it too turned out to have an incomprehensible MIDI spec, and it took us at least half an hour to figure out how to change the channel on which it received notes.

In a modern context, it would have been the work of moments to shove a compressor plug-in over the DX piano, but that wasn't an option for home studios of the '80s. Happily, Notator actually features quite a comprehensive set of real-time MIDI processors and effects, so although some manual editing of note velocity was necessary, adding some MIDI compression took care of the worst of it. Meanwhile, it turned out that the one thing that the Juno would reliably respond to was program change messages, so we were able to program a couple of slightly different patches for different sections of the song. Once we'd fired up the REV7s and added swathes of cheap reverb to everything, it was beginning to come back together again, though we eventually ended up keeping a few of the more incidental GM sounds, simply because they sounded so gloriously '80s.

With our MIDI arrangement almost complete, it was time to think about recording some audio. We had a tape machine, we had tape, we had looms and isopropyl alcohol for cleaning the heads. What could possibly go wrong? Pretty much everything, as it turned out, as Dave recalls: "We started with a Foxtex M80 quarter-inch eight-track, found that it wasn't working properly, went to a B16 that we also soon found didn't work properly, nearly bought a Tascam 38 on eBay, tried to borrow an R8, and then fixed the M80 that we started with. Of course, it didn't stay fixed: tracks one and eight ceased recording as soon as we got everything plumbed in, but we decided this was not too atypical of the '80s home-studio experience, and agreed to press on regardless. With track eight inoperative, track seven became our new 'edge track' to be used for SMPTE timecode, which left five working tracks for audio."

If there was one positive that surprised us about the tape experience, though, it was the resilience of the synchroniser. Despite the indignities to which our timecode track was subjected by the tape machine, we never once lost sync, and the Atari would resume MIDI playback almost as soon as the recorder got up to speed.

We had always planned to run the MIDI instruments live at mixdown, but even then, given that we had hoped to record doubled lead and backing vocals, at least two electric guitar tracks, acoustic guitar and fretless bass, it wasn't obvious how we were going to fit everything in. Putting off that dilemma for the time being, we started by recording electric guitar.

"If pushed to think of a typically '80s guitar sound, many of us would probably identify 'clean, compressed, 'clicky' DI'd Strat' as a candidate. The 'clean' setting of a Tom Scholtz Rockman box is probably responsible for popularising this sound, but a nicer version of it, with less noise, can be achieved with a suitably squashy compressor pedal straight into a DI box. For this we used a '70s Strat, on the 'inbetween' setting between bridge and middle pickups, through a clone of the MXR Dynacomp compressor, which kills the transients to just the right extent, feeding a 1980 BSS active DI box with a very useful 8kHz low-pass filter setting.

"Even in the smaller home and project studios of the '80s, most players generally preferred to mic an amp for any distorted guitar parts, whenever possible. Acceptable DI solutions simply weren't available, perhaps until the advent of the Tom Scholtz Rockman - although some might argue the assertion that the Rockman's distorted sound approached acceptability!

"A slightly later, and better, DI solution was the original SansAmp pedal. I discovered you can make one of those sound a bit like a Rockman if you dial it in really badly, but you wouldn't have thanked me for treading all over the track with that, and in the end I settled for something fairly generically amp-like from the SansAmp using its Mesa Boogie settings. The same sound was used for the crunchy chording and the second half of the solo."

Dave's electric guitar parts occupied two of our five available tape tracks, albeit with occasional gaps. Next up, we decided, would be acoustic guitar: our first foray into recording an acoustic source. These days, we are spoilt for choice regarding studio microphones, but prior to the influx of affordable Chinese-built capacitor mics in the '90s, it was a very different state of affairs, and it would be a rare home studio that owned a range of mics for different purposes. Much more likely was that a single half-decent mic would be pressed into service on any source, and we decided that the only mic available to our time-capsule home studio would be an AKG C451 with the CK1 cardioid capsule. This combination is still widely used today on acoustic guitar, but probably isn't something most of us would reach for on male or female vocals!

The dynamic range available on even the most basic of today's computer-based recording systems typically exceeds 100dB, meaning that you can pretty much leave as much headroom as you like when recording quiet or dynamic instruments; the limitation on the dynamic range in the recording will be due to ambient noise in the recording space rather than system noise. Things are a bit different when recording to narrow-band tape on a poorly maintained, 30-year-old eight-track machine, and it took some juggling of gain pots and a firm strumming hand to get a decent level to tape. The C451's fierce high-frequency lift, which can be a hindrance with digital recording, helped to combat the inevitable loss of top end that comes with the recording format.

In 1985, as now, the most important part of any pop track was the vocals, but the number of products targeted specially at recording and mixing the human voice has mushroomed since the '80s. As well as the aforementioned 'voice channels', most of us probably have access to pitch- and time-correction, de-essing and other vocal-oriented effects, not to mention unlimited recording tracks with advanced editing and comping tools. So it was an interesting experience recording vocals to a single track, with options limited only to 'keep it' and 'do it again'. In practice, though, we quickly got used to the approach of working through the song, recording a verse at a time and performing drop-ins to correct any dodgy lines. Once again, it meant committing to a take straight away rather than putting off the decision until mixdown, but although my final vocal performance had plenty of ropey bits in it, that was more to do with lack of time and an over-ambitious harmony part than with the limitations of the recording process.

The '80s was arguably the golden age of the pop duet - who can forget such gems as 'Don't Give Up', 'Up Where We Belong' or, er, 'Especially For You'? - and our song lent itself fairly naturally to this treatment. The two vocalists were, however, recorded separately: not an uncommon procedure when a home studio may perhaps have had only one good microphone.

Nell Glasper, Sound On Sound's Production Editor, is a fine singer of traditional and contemporary folk music, but soon locked on to the necessary 'detached' vocal style required of the female lead here. With Nell's main vocal down, we were faced with a dilemma. We wanted the female vocal to be doubled, but we had run out of working tracks! So we took the bold decision to erase the acoustic guitar, which seemed a bit less crucial to the arrangement once the vocals were in place. It was a decision we never regretted: Nell's accurate double perfectly captured the full '80s 'disinterested vocalist' effect. The only place where the acoustic guitar was really missed was in the middle eight, where one of Dave's electric parts did not play, so with some nifty drop-in work we were able to reincorporate a new take just for this section.

That chorus-y double-tracked sound is very characteristic of '80s pop production, so we decided to try the same trick on the bass. Elsewhere in the song, the synth bass was a pretty dull pedal-note affair, but we'd come up with quite a nice melodic line for the middle eight, and there was a gap in the male vocal part at this point, so we supplied Chris Korff with a fretless MusicMan and set him to work.

With a bit of imagination, it's surprising how much you can fit on to a handful of tape tracks, and had we been working on the song for longer, I'm sure we'd have come up with more ideas to squeeze into unused corners of the arrangement. With a deadline looming, though, we decided it was time to integrate the one piece of '80s music technology we hadn't yet exploited: FX's Akai S950 sampler. Although we had a decent supply of 3.5-inch floppy disks, what we didn't have was the factory library or indeed any sounds for the unit, so Reviews Editor David Glasper was assigned the unenviable task of programming something from scratch. There wasn't a pressing need for additional musical elements in the song, so he chose to reinforce the lyrical theme by sampling and manipulating some classic '80s video-game sounds (see box), which we integrated mainly into the fade at the end.

Finally, it was News Editor Will Betts who supplied the finishing touches to our track, with a couple of magnificent orchestral stabs and a virtuoso General MIDI helicopter solo - anachronistic perhaps, but very, very '80s in sound. It was time to mix.

The very idea of a separate mix stage has become unfamiliar in the DAW world, and it's perhaps in mixing above all that the difference between old and new emerged most clearly. In a computer-based system we have complete freedom to EQ, compress, effect and automate every last shake of a tambourine or pluck of a string, and often enough, we start to do this as soon as the tracks are laid down, safe in the knowledge that everything can be ripped out and changed at a moment's notice.

In the DAW world, everything we do at the mix can be automated, edited and stored for later recall. Mixing on an analogue desk is much more like a musical performance, and the same recording of the same song will come out differently every time. Luckily, we had plenty of hands available to man the faders, and our MIDI arrangement was sufficiently developed that few fader moves were required in that department. However, two of the TR707 sounds we were using - the handclaps and the tambourine - shared an output, and one seemed much louder than the other, so Nell was deployed to adjust the tiny onboard slider that controlled their level. As the sources most in need of fader riding, the vocals each got their own operator, while Chris Korff was deputed to add spacey delays to the video-game samples.

After a number of rehearsal passes, we had come up with a fairly detailed map of the song with levels for each section. Even then, we had to do four or five 'takes' of the final mix before we got one that we were satisfied with; the difference between something's sitting at the right level and its being noticeably too loud or quiet could be as little as a couple of millimetres of fader travel. Knowing that the song would eventually need to end up in the digital domain if anyone was to hear it, we cheated slightly and recorded the master to a Mac as a 24-bit WAV, as well as to DAT and cassette, for the sake of authenticity. The latter sounded a lot better than a lo-res MP3.

Between us, the SOS staff who participated in this project can look back on more than a century of home-studio recording experience. Yet apart from Dave Lockwood, none of us have first-hand experience of the studio world into which SOS was born three decades ago. If there's one thing that our attempt to recreate this world brought home, it's that music technology has always been the tail that wags the dog. Working with an '80s studio makes crystal clear why so much '80s music sounds as it does! Machine-like, hard-quantised backing tracks; arrangements given life by MIDI mutes rather than human performance variation; unconvincing synthetic recreations of real instruments; digital reverb as a sticking plaster to cover up all sorts of evils: so many of the characteristic qualities of '80s records came about not because they were intrinsically desirable, but because they represented the path of least resistance given the technology of the time. It would have been far harder to use our '80s home studio to make music that didn't recall that decade.

Today's music technology is infinitely more powerful, and the ways in which it shapes our music are perhaps more subtle - but are we any closer to being its masters than was the case 30 years ago? Or does today's technology merely push us in different directions?

Thanks to FX Rentals for supplying the instruments used in this feature.

I started my recording career with a Studer 24-track in a pro studio, but I did my fair share of home and project studio recording, too. In my experience, a typical mid-to-late '80s home studio would be based around a multitrack tape machine - quarter-inch eight-track, half-inch eight-track or half-inch 16-track, according to budget - and a computer running MIDI sequencing software. The computer would almost certainly have been an Atari 1040, with the most popular sequencers being Steinberg's Pro 24 and later Cubase, and Creator/Notator from C-Lab (later called E-Magic). A SMPTE timecode synchroniser would have been used to lock the sequencer to the tape machine, so all the MIDI sources could remain 'live' through to the final mix.

In the earlier years, synths would often only run a single polyphonic voice, so you would need a few of them for a full production, leading to the popularity of keyboardless voice modules like Yamaha's TX7, which offered almost all the functionality of a DX7 in a compact format. Samplers, like Akai's popular S900, were inherently multitimbral instruments with a variety of applications and would often be used in place of a dedicated drum machine, with every voice split out to its own output.

Combining these sources at mixdown called for an inconveniently large number of inputs, often with every aux and tape monitor return pressed into service alongside the input channels. Affordable digital reverb from companies like Alesis and Yamaha added still further to the pressure on line inputs - somehow there were never enough.

In a typical setup of this period, there would be a modest amount of outboard processing: perhaps one decent stereo compressor for mixdown, which would also have been used for tracking vocals. Mic choice too would often have been quite limited: Shure SM58s and 57s were often pressed into service, or AKG's C1000 and C451 capacitor models. AKG's D190 and the rather nicer D202 and D224 were certainly around in home studios, and maybe something affordable from Sennheiser or Beyer, but if you had a large-diaphragm, multi-pattern model back then, you'd be considered practically a pro.

Mixdown would generally have been to quarter-inch analogue tape or possibly even cassette, for those who had already spent too much on the rest of their system. The DAT format of compact digital audio cassettes was still a few years off, but for the really flush there was always the option of a Sony PCM F1 and Betamax recorder. The F1 was a digital converter that stored its bitstream as a video signal onto Betamax video cassettes. It was a bit finnicky to use, but sounded OK, and I mastered a fair few albums with it until DAT came along. Dave Lockwood

When we were drawing up our list of must-have '80s equipment, an Akai sampler was near the top. This really was cutting-edge technology at the time: when Akai released the S612 in 1985 it was arguably the first sampler that could be considered anywhere near affordable. It came with 128kB of internal memory, with which you could sample up to eight seconds of audio at the lowest 4kHz resolution (a single second was available at the maximum 32kHz), and an optional 2.8-inch Quick Disk drive was necessary if you wished to preserve your samples for posterity. Still, you could hope to own one without selling a grandmother (an Emulator 2 cost approximately nine grandmothers back in 1985) and sampling suddenly became feasible for the project studio owner.

S612s are pretty thin on the ground these days, and 2.8-inch QD drives even more so, and when we discovered that the extremely helpful people at FX Rentals had an S950 we could borrow, we realised we had to compromise slightly on period authenticity. Nonetheless, the S950 is not too far removed from its forebear, being an updated version of 1986's S900. It's still a 12-bit machine, but offers sampling rates of up to 48kHz and an internal memory that's expandable to a whopping 2.5MB. It also has a standard 3.5-inch floppy drive, and as we never throw anything away at SOS, finding a few disks for it wasn't going to be a problem.

So, faced with a song with lyrics about video games and an insatiable appetite for as many corny '80s clichés as we could throw at it, it was obviously necessary to sample some eight-bit computer-game sounds and use them to decorate the song's intro and outro. And as the onetime owner of an S950, it fell to me to play sampler tech...

I began by firing up the machine and staring at it uncomprehendingly for a while. It's been at least 13 years since I last used an S950, and I quickly realised I'd forgotten everything I ever knew about programming one. Actually, I didn't stare uncomprehendingly for long, because the 950 is a very simple machine to use, and has a number of helpful instructions printed on its front panel. With these instructions as a crib and a little exploration of the menus, one can begin recording and editing samples almost immediately. In fact the only time I needed to refer to the manual I'd so diligently downloaded beforehand was to find out how to add another keygroup. (The manual, by the way, is charmingly written and is a refreshing change from the usual dodgy translation from the Japanese typical of the period. Bravo to whoever was responsible.)

I didn't attempt any Fatboy Slim-style Akai stunt programming, and simply sampled a dozen or so sounds from a popular eight-bit computer game (also, coincidentally, released in 1985), which I reckoned gave a fair selection of the kind of crunches, bloops and beeooow sounds we were looking for. I then set about programming them across the keyboard.

This was where the complete lack of a graphical interface really made itself felt. Instead of drawing or dragging anything with a mouse, it was a case of manually selecting a sample, setting its pitch and then setting a maximum pitch range, all with a grid of buttons and a data wheel. Easy enough with a dozen samples, perhaps, but I hate to think how long it would have taken to program a properly multisampled instrument. (That said, I used to own a Roland S550 with a green and black CRT monitor that gave you a waveform display, amongst other things, and that used to give me actual nightmares, so perhaps a GUI isn't everything.)

Once I'd relearned how it worked, I actually began to enjoy using the S950. It certainly has a sound of its own, and once you get used to the lack of any modern editing facilities, it becomes rather fun to use. It also produces far more unexpected results than its modern, computer-based equivalents, perhaps as a result of my less-than-expert programming, and I found that firing random notes at it from an Arturia Beatstep Pro created all sorts of interesting rhythmic effects, which I in turn sampled back into my computer (taking a fraction of the time to do so, I might add, and with no worries about running out of memory). It also has an endearing habit of defaulting to 'Omni On' (in fact the front panel's 'On' and 'Off' buttons seem to exist solely to adjust this control), which caused all sorts of bonkers noises to issue forth every time we switched it on.

For conventional sampling work there's very little to recommend a sampler of this sort, but for creative serendipity or sheer grit they're hard to beat. We've certainly gained a lot in terms of usability and speed in the last 30 years, but perhaps we've also lost something of what made sampling so interesting in the first place. David Glasper

Point your browser at http://sosm.ag/nov15media to hear our '80s mix in all its glory - including versions mastered to DAT, cassette and MP3!

To mark our 30th anniversary, the SOS team chose some highlights from the history of modern music technology. Click the link to download our Timeline Of Tech! timeline-tech-1115.pdf

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Recreating The '80s Home Studio Experience

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To celebrate the magazine's 30th birthday, the SOS team take a trip back in time, finding out what it was really like to work in a mid-'80s home studio. Nostalgia is a kind of selective memory.

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If you are going to add to the list please use the same format and insert in chronological order. check this for a short overview about "who created the synthesizer" http://www.youtube.com/watch?v=C7zRUSOhgKM

The Synth List - Formating: Year - Manufacturer - Model - First at what..?

1837 - C.G. Page (Salem. Mass) - first to produce electronically generated sound (not necessarily associated with a musical instrument). After inventing the Volta in 1800 (an early battery), in 1837 Page was doing experiments with coils and realized when certain coils were attached to a batter they omitted a ringing sound. While he initially thought the ring came from the electrical current was interrupted (battery disconnected), what was actually taking place was the induction through the coils was causing them to vibrate. via Peter Grenader

1885 - Person and Ernst Lorenz -'Elektrisches Musikinstrument' - the first musical instrument designed to produce electrically generated sound. It used electronic vibrations to drive an electromagnet that were connected to resonating boards, which translated these vibrations to sound. via Peter Grenader

1897 - Taddaeus Cahill - Telharmonium - electromechanical instrument.

1936 - Oskar Sala - Mixturtrautonium - first electronic instrument using Subharmonic synthesis, not a "synth".

X-1905 Helmholtz Resonator (Human Voice "Emulation") http://www.sequencer.de/blog/helmholtz-resonator-1905/20116

1938-42 Oscs, Filters, Envs, LFO (Mech) - no CV - Hammond Novachord

1939 - Homer Dudley invents the Parallel Bandpass Vocoder (VODER) - A manually key operated speech synthesizer

1940 - Homer Dudley invents the The Voder speech synthesizer - A device which used the human voice and an artificial voice to produce a composite. Both were researched as a way to transmit speech over copper wires (id est, telephone lines)

1948 - Hugh LeCaine - Electronic Sackbut - First voltage-controlled synthesizer

1948 - Dr. Raymond Scott - Wall of Sound - First polyphonic Sequencing Workstation (electromechanical) and the Electronum - first sequencer.

1950 - CSIR - Mk 1 - The first known use of a digital computer for the purpose playing music // Alan Turing sequences melodies with computers http://www.sequencer.de/synthesizer/viewtopic.php?f=70&t=117493&p=1377621#p1377621

1953 Harald Bode - it had: OSCs, Filter - Ringmod - Reverb - finally named: Melochord

1953 RCA-Synthesizer: Herbart Belar, Harry Olson - RCA Synth Mk1, 2 (rec. auf Vinyl) ///1955 mk1 / 1957 mk2 made mk2 used tape instead of vinyl disc

1956 - Louie and Bebe Barron - Produced the first all-electronic musical score for a major motion picture - MGM's 'Forbidden Planet'

1957 - Max V. Mathews at Bell Labs - MUSIC - the first digital synthesizer. Technically, it was a computer program, though it set the stage for every digital synthesizer that proceeded it.

1961 - Harald Bode his studio modular lab was- very much like the first synth made by HB except oscillators (tapes were used) - but it had filters etc.

1964-65 (unsure) - Buchla - model 100 modular - 1st "modern" modular synth

1967 - Moog - Moog modular synthesizer I, II & III - 1st commercial modular synth.

1969 - EMS - Synthi VCS-3 - first non-modular mini-synth

1970 - Moog - Minimoog - 1st Mono Synth with keys (non-modular)

1971 - Tonus/ARP - Soloist - 1st preset mono synth

1971 - John Chowning - developed FM synthesis using the MUSIC-IV language (source), a direct descendent of Mathew's MUSIC program. FM synthesis was later licensed by Yamaha, and used in popular synths such as the DX7.

1971 - Buchla - 500 - micro-controlled polyphonic analogue in 1971, it was also programmable as you could save patches to floppy.

1972 - Triadex Muse - first of many horrible sounding digital synth/seq workstation thingies

1973 - Coupland Digital Music Synthesizer - First Digital (Triadex beat it?) Update via Peter Grenader: "No time to read through all these posts to see if it's come up yet, but the Coupland was vaporwear...it never existed. I met Mark Vail, who's now a friend, by writing him a letter informing him that his story about the Coupland in his Vintage Synthesizers book (GREAT book) which mentioned it's only recorded showing was at the AES show in LA in 1978 was a farce. I was there - at their booth and their suite in the Hilton where the instrument was said to be. I was there on the first day, I was there on the last day. The only thing they had was a small model - about six inches across, sitting on a table. The booth was amazing - this radial orb multiple people could sit in, with a cover that came over each person which played what I remembered was a very impressive demo which swirled around four speakers inside the box. I, and everyone else, were blown away. They kept saying...'it will be here tomorrow, it'll be here tomorrow'...so I showed up the last day just to see it, figuring by the then it would have arrived...it didn't. I did see the frst Synclavier at that show however. Their suite was across the hall from the Coupland folk. That completely kicked the crap out of everything else shown that year."

1973 - NED - Synclavier - first digital synth

1974 - Roland - SH3a - first commercial additive synth

1974 - RMI - Harmonic Synthesizer - first commercial additive synth

1976 - Yamaha - CS80 - first synth with poly aftertouch = polypressure

1976 - PPG - PPG 1003 sonic carrier - 1st programmable mono/duo synth (this, along with the model 1020, might have been the 1st synths to use DCOs as well)

1977 (late) - Oberheim - OB1 - 1st commercial programmable mono synth

1978 (late) - PPG - Wavecomputer 360 - 1st wavetable synth

1978 - Sequential Circuits - microprocessor control the SCI Prophet10 (briefly) and the P5 --- again based on existing E-mu tech stuff

1979 - Fairlight CMI - First Sampler, First Workstation

1982 - Sequential Circuits - Prophet600 / First Midi Synthesizer (though some argue the Prophet 5 rev 3.2 is pre-MIDI MIDI)

1983 - Yamaha - DX7 - Digital takes over, FM goes mainstream

1983 - OSC - OSCar - First real-time additive with analog filters

1984 - Sequential Circuits - SixTrak - first multitimbral

1985 - Casio - CZ101 - First battery-powered all digital mini-synth

1989 - Emu Systems - Proteus - First dedicated ROMpler

1994 - Yamaha - VL1 - first physical modelling synth

1995 - Clavia - Nord Lead - 1st Virtual Analog (VA)

1996 - Rubberduck - still not the first softsynth but came before Seer Systems Reality.

1996 - Steinberg - VST - Ok not a synth but enabled a lot to be written as plug-ins and used simultaneously

1997 - Seer Systems - Reality - First Modular Soft Synth

2912 - KalQuestoTron - the first genetically engineered synth. Each cell is an oscillator, filter, and neural sequencer. Can be delivered via injection to always play 'hold music' in your head.

Related Links - Synthesizer History

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If you are going to add to the list please use the same format and insert in chronological order. check this for a short overview about "who created the synthesizer" http://www.youtube.com/watch?v=C7zRUSOhgKM 1837 - C.G. Page (Salem. Mass) - first to produce electronically generated sound (not necessarily associated with a musical instrument).

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Synthesizer and electronic music news, synth and music software reviews and more!

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At first glance, the QUN Pocket Synthesizer by Nunomo (new-no-moh) doesn't' look like much, but closer inspection reveals quite a powerful little device.

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This page contains links to every part of Gordon Reid's hugely popular SYNTH SECRETS series which ran for over five years in Sound On Sound magazine (May 1999 to July 2004).

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Sound on Sound

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As the nails are being hammered firmly into the coffin of competitive loudness processing, we consider the implications for those who make, mix and master music.

In a surprising announcement made at last Autumn's AES convention in New York, the well-known American mastering engineer Bob Katz declared in a press release that "The loudness wars are over." That's quite a provocative statement - but while the reality is probably not quite as straightforward as Katz would have us believe (especially outside the USA), there are good grounds to think he may be proved right over the next few years. In essence, the idea is that if all music is played back at the same perceived volume, there's no longer an incentive for mix or mastering engineers to compete in these 'loudness wars'. Katz's declaration of victory is rooted in the recent adoption by the audio and broadcast industries of a new standard measure of loudness and, more recently still, the inclusion of automatic loudness-normalisation facilities in both broadcast and consumer playback systems.

In this article, I'll explain what the new standards entail, and explore what the practical implications of all this will be for the way artists, mixing and mastering engineers - from bedroom producers publishing their tracks online to full-time music-industry and broadcast professionals - create and shape music in the years to come. Some new technologies are involved and some new terminology too, so I'll also explore those elements, as well as suggesting ways of moving forward in the brave new world of loudness normalisation.

Anyone who has contemplated putting their own music onto a CD or online, even if only for family and friends, will be all too aware of the 'Loudness Wars'. This long-standing practice of competing to make one record sound loud in relation to others is usually claimed to be the consequence of an observation made in the 1950s that people tended to play the louder-cut records in jukeboxes more often. Thankfully, there's a physical limit to how loud a vinyl record can be cut without making it unplayable, so even the loudest-cut records managed to retain quite reasonable dynamics. Unfortunately, digital recording removed such constraints - a CD, for example, is playable regardless of the amplitude of the encoded audio data - and that simple technical freedom facilitated the 'war' that has been raging with (arguably) ever more musically destructive power over the last 30 years.

The audible consequence has been that the 'volume' of pop, rock and other music recorded and released on commercial CDs has risen steadily since the late 1980s, with a corresponding reduction in dynamics and, in many cases, a trend towards a more aggressive and fatiguing sound character - all in an attempt to make each track as loud as or louder than the perceived competition. It's bewildering to think that the audio format that offered the greatest dynamic range potential ever made available to the consumer is now routinely used to store music deliberately processed with the least possible dynamic range in the history of recorded music!

If Bob Katz is proved right, this current fashion to 'hyper-compress' music may well be relaxed or even reversed in the months and years to come, and the application of compression and limiting will revert to being a purely artistic and musical decision, rather than being an essential process in making a competitive product.

The key to Katz's claim is an ongoing industry shift into a 'loudness normalisation realm', in which the replay level of individual tracks is adjusted automatically to ensure they all have the same overall perceived loudness. This already happens to some extent on commercial broadcast radio, for example, but the new technology makes it practical to employ it in Internet music-streaming services and, crucially, personal music players too. Within a loudness-normalisation environment, it becomes impossible to make any one track appear to sound louder, overall, than any other, so mastering to maximise loudness inherently becomes completely futile.

A second aspect to this paradigm shift is that the kind of hyper-compressed material which the loudness wars encourage ends up sounding very flat, lifeless and even boring when compared with tracks which retain some musical dynamics - and the loudness-normalisation world really encourages and accommodates the use of musical dynamics.

So, in loudness-normalised environments - which is likely to include the vast majority of music replay systems within the next year or two - the use of heavy compression and limiting to make tracks sound loud will no longer work. Each track is balanced automatically to have the same overall loudness as every other track, and hyper-compressed material actually ends up sounding flat, weak and uninteresting compared with more naturally dynamic material.

Katz argues that the universal adoption of the loudness-normalisation paradigm in the consumer market will inevitably create a strong disincentive for the use of hyper-compression, and instead encourage music creators to once again mix and master their music to retain musical dynamics and transients. If this proves to be the case then the loudness wars may indeed be over, at last... but it will be a while before the last skirmishes are fought!

The current trend is all about trying to make a CD or download sound as loud as possible compared with other commercial material. Yet listening loudness is easily determined by the user via the volume control - so this really is just about competitive loudness, which is largely destructive and damaging.

In the all-analogue days, mixing consoles, tape machines, vinyl discs and other consumer replay media all employed a nominal 'reference level' - essentially the average programme signal level. Above this reference level, an unmetered space called 'headroom' was able to accommodate musical peaks without clipping. This arrangement allowed different material to be recorded and played with similar average loudness levels, whilst retaining the ability to accommodate musical dynamics too.

With the move into digital audio, the converters in early CD mastering and playback systems weren't as good as they really should have been, and to maximise audio quality, the signal levels had to peak close to 0dBFS (ie. the maximum digital peak level). Consequently, the 'reference level' effectively became the clipping level, and the notion of headroom fell by the wayside, becoming this war's first casualty!

Music producers (in the broadest sense of the term) quickly realised that they could employ techniques to squeeze as much of the audio signal as possible up towards the digital system's maximum peak level, simply because the CD format didn't complain like vinyl cutters did - and now many people seem to think that this is the primary role of a mastering engineer! The typical way that a track is made to sound loud on a CD (or as a download) is by employing heavy limiting and compression, often combined with some spectral shaping, to increase the average energy level and minimise the crest factor, or peak-to-average ratio. These techniques work because the way we perceive loudness is essentially based on the average energy level of a track; the higher the average energy level, the louder the track will sound. The technical term is 'peak normalisation': raising the level of the signal so that its peaks hit a defined maximum level. The flip side of this coin is that we lose the headroom margin, so there is no longer any room for musical dynamics. We can't raise the peak level any further, only the average level through limiting. In extreme cases, the material will become hyper-compressed.

Hyper-compression isn't an intrinsically bad thing: it has its place, just like any other musical effect, and has become an integral part of the sound character of certain musical genres. That's fair enough, because it's used in a musically informed way. But where it's being used purely to try to make a track louder than the competition, it can be quite damaging, either in the strict technical sense (through clipping, pumping and other distortions), or by compromising the musicality to some degree. The extent of the damage can vary from barely noticeable to blatant, depending on the skill of the mastering engineer and the requirements of the client. But, as a general principle, moving away from this long-established but characteristically destructive peak-normalisation practice would be a good thing from many perspectives. That's certainly what the TV industry felt a few years ago - and that's where the origins of the loudness-normalisation technology came from.

At the heart of the loudness-normalisation movement lies something mysteriously called ITU-R BS.1770, which was developed in response to years of viewer complaints to television companies all around the world about overly loud adverts. (If there's anyone who hasn't noticed that phenomenon, I've yet to meet them!) So, at this juncture, I need to take you on a short diversion into the dominion of the broadcast industry...

Historically, trained broadcast sound operators worked in 'continuity suites' or 'master control' rooms, using their ears and audio skills (with guidance from their console meters) to balance and control the broadcast sound levels, instinctively matching the perceived loudness of different contributions and programmes. This was, therefore, a 'loudness normalised' environment, and it worked very well indeed. Consumer research has revealed that most people ignore modest loudness variations within a 'comfort zone' of 2.4dB above and 5.4dB below their preferred volume setting, which is easy to maintain with trained sound technicians. I can remember that when my father bought our first colour TV, way back in 1970, we almost never had to adjust the TV's volume control, such was the broadcast consistency of the time.

However, skilled staff are expensive and so the broadcasters gradually swapped staff for automated systems, and these simply played scheduled programmes directly to air. Audio level adjustments would only be made if the material exceeded a defined Permitted Maximum Level (PML). Of course, it didn't take the advertisers long to realise that they could exploit these systems by employing heavy compression and limiting to make their adverts sound much louder than other adverts or the enveloping programmes, while keeping peak levels at the PML limit. Their aim, of course, was to make people take more notice of their ads and thus (hopefully) increase sales.

The TV industry had, in effect, drifted into a peak-normalisation environment and, sadly, we're all familiar with the consequence, through the constant need to adjust the TV volume between programmes and adverts, as well as between different channels. The loudest adverts could easily be 4-8 dB louder - far more than the typical tolerance range - than the adjacent programming, and hence the flood of complaints to the broadcasters.

Various attempts over a decade or more to introduce industry self-regulation to address inconsistent TV loudness failed. America even went as far to legislate, with its Commercial Advertisement Loudness Mitigation (CALM) Act. Meanwhile, research was well underway to find a technical solution, and that culminated with the International Telecommunication Union's (ITU) release of ITU-R BS.1770. This is a technical recommendation which defines a 'loudness metering' algorithm (see side boxes) which provides a standardised, objective and reliable means of measuring, comparing and adjusting programme loudness.

Note that this is about loudness as perceived by human beings, and not simply 'level'. In other words, the ITU-R 'Loudness Meter' algorithm is essentially an 'electronic ear' that perceives audio loudness in much the same way as the average human listener, and it has proved to be very accurate and very reliable. The primary intention behind its development was to enable broadcast automation systems to measure the perceived loudness of supplied programmes and adverts before they were broadcast, so that their loudness levels could be matched in a similar way to that achieved by those trained continuity operators 30 years ago. Plus ça change!

The ITU-R BS.1770 standard was formally introduced back in 2006, and has been adopted by most of the world's HDTV broadcasters. There have been a few updates along the way, and regional broadcast standards bodies have issued their own application guidelines to suit local practices. For example, in Europe the EBU (European Broadcasting Union) version is called R-128, while the USA's variant is known as ATSC's A/85. In essence, though, they all do the same thing, work the same way, and have virtually identical specifications.

In my native UK, all high-definition TV programming on the BBC, ITV, Channel 4 and Sky now complies fully with the R-128 specifications through a voluntary agreement, and the benefits are quite audible. Adverts no longer deafen the audience, changing channels doesn't require an adjustment to the volume, and the viewer enjoys far more consistent programme levels generally. The introduction of loudness-normalisation has been a positive improvement to the TV viewing experience. (If you're still wondering what all this has to do with releasing music on CDs, stick with me - I am getting there, honest!) The BS.1770 recommendations have been implemented on HDTV channels first because they're all digital, and thus have vast dynamic-range capability, very low noise-floors, and absolute level calibration. This means they can accommodate sensible headroom margins without compromising sound quality, and a digital reference level means the same to everyone.

An important fact to note about loudness normalisation is that it is nothing more than a static volume adjustment for each programme, based on an assessment of the programme's qualitative loudness level, measured over its full duration. That last point is important. The loudness value is determined across the whole programme duration, whether it's a 30-second commercial or a two-hour feature film, and not moment to moment. Once the loudness value is determined, the replay level can be adjusted for that programme or advert, so that its loudness conforms to the defined 'target loudness level' of the broadcast system.

The programme's internal dynamics aren't changed in any way; it can still have quiet moments and loud explosions, if that's what the action calls for. Neither is the programme's spectral balance changed - there's no radio-style multi-band processing involved. Loudness normalisation just does what most listeners do instinctively with their remote control; it adjusts the replay level to a comfortable volume and maintains that automatically between different programmes or channels.

However, news programmes or chat shows typically have a fairly small dynamic range, whereas a feature film might have a massive dynamic range. The chosen target loudness level has to accommodate both, and that means building in an adequate headroom margin. There's an old-fashioned concept! The BS.1770 recommendation calls for a target loudness level for HDTV of around -23LUFS (Loudness Units relative to Full Scale, see box), although different regional implementations vary slightly (for instance, the ATSC spec is 1LU lower) and programmes being mixed live are allowed to deviate slightly from the target loudness (typically up to ±1LU).

What this means in practice is that the typical volume of normal dialogue (which makes up the bulk of most programmes) will have roughly the same perceived loudness, whether in a news bulletin or a feature film, and that will typically be about -23LUFS. However, while the signal peaks in a news programme might reach only 4dB higher than the average dialogue level, the occasional explosions in a feature film might reach a whopping 20dB higher. The beauty of the loudness-normalisation paradigm is that it doesn't mind, because the target loudness level has been chosen to provide sufficient headroom to cope with such peaks. Importantly, the listener perceives both programmes as having the same overall loudness, despite their very different dynamic ranges.

So, the TV world, finally getting its house in order, is delivering programming with far more consistent overall loudness. Hurrah! But you're still wondering what that's got to do with the way you mix your music. We're getting closer to that, but our detour must first lead us through the world of broadcast radio and Internet streaming services...

TV isn't the only broadcast sector that has a loudness problem. Reflecting the 'jukebox theory' I mentioned earlier, popular music radio stations have long been trying to make their output sound louder than their competitors', in the hope that listeners would choose 'the loudest station on the dial'. Like the CD, though, analogue radio transmitters have a finite maximum level capability, and so complex, automatic, multi-band compression and limiting techniques are routinely employed to maximise broadcast loudness.

So, here we have another example of a peak-normalisation paradigm, but in this case, the broadcast processing generically known as Optimod (although that's a proprietary brand of the Orban company) alters the source music's spectral balance and dynamics, and often does so quite dramatically. This methodology also causes a volume disparity when listeners retune from more conservatively processed classical music or speech stations. Try switching from BBC Radio 4 to Radio 1 to see what I mean!

As the industry moves further into digital radio broadcasting and Internet streaming, with hundreds of stations accessible at the touch of a button, there's a growing consumer pressure to adopt the loudness-normalisation model instead, and the various audio industry bodies are currently working to ratify a standard. At the time of writing, the optimum target loudness value for broadcast radio is still being debated (although it's likely to end up somewhere between -15 and -23LUFS), and I'm not aware of any broadcast radio station that has taken the leap into loudness-normalisation yet. They're probably all awaiting the release of an industry-wide 'standard' to guarantee a level playing field!

The Internet music streaming service Spotify has been using a form of loudness normalisation from the day it was launched, and Apple have recently made the same leap of faith with their new iTunes Radio music streaming service (which at the time of writing is only available in the USA). Consistent loudness is maintained, regardless of the musical genre being streamed, by using iTunes' Sound Check function (see box), and although Apple haven't stated publicly what target loudness value its Sound Check function currently imposes, Bob Katz claims it to be an entirely credible -16.5 LUFS. The adoption of a loudness-normalised paradigm in such a widely used music streaming service is going to be very influential, and could potentially have a massive influence not just on the rest of the music streaming industry, but on the music creators as well, simply because the consumer will become used to loudness-normalised material and the dynamic range capabilities presented in that paradigm. This is the catalyst for Katz's recent press release claiming the loudness war is over.

So, after meandering around the broadcast industry's working practices, we arrive at the crux of the whole loudness-normalisation subject. A significant proportion of domestic music consumption is via broadcast radio and Internet streaming, and both of those outlets are adopting loudness normalisation. Most popular personal music players also have loudness-normalisation options, like the Sound Check function on Apple's iPods and iPhones, and on computers when using their iTunes playback software, to maintain consistent loudness when music is played in shuffle or playlist modes. Loudness normalisation means there is no advantage in producing 'loud' CDs or downloads, because the perceived loudness will be the same for everything. The really sobering aspect of loudness normalisation is that very heavily compressed music ends up sounding dull and feeble compared with more naturally dynamic material.

Before the panic sets in, I think it's important to appreciate that we're not facing an overnight step change to music production techniques as we know them! Moreover, no-one is forcing anybody to change the way they mix and master their music; if you like and want to hyper-compress your tracks, you still can. However, you also now have the option to employ musical dynamics without worrying about the CD 'not being loud enough'.

Assuming loudness normalisation catches on - and there are certainly some areas where it may not, such as in the club-music scene - the evolution of production and mastering styles to relax hyper-compression and embrace dynamics will be a gradual process. It will take time for people to become familiar with the new metering technology, and working practices will evolve and spread slowly through the industry. Consumers' acceptance and expectations will also develop slowly, and a new generation of music players may be required before loudness normalisation becomes completely practical.

In that respect, this 'loudness revolution' is really no different from the many other radical changes we've already witnessed - and survived - in the music industry, such as high-fidelity microgroove records replacing shellac 78s in the 1950s, or the introduction of stereo in the '60s, of multitracking and overdubbing in the '70s, of synthesizers in the '80s, of music sequencers in the '90s, and of the Digital Audio Workstation at the turn of this century. All of these technologies (and more besides) have both shaped the way music has been made, and influenced the way it actually sounds. The loudness-normalisation paradigm will impart its own influence on the sound of music too, the most important part of which will be to reinvigorate the creative use of dynamics.

The important thing for SOS readers right now is to be aware of the forces driving these potential changes to the long-established working practices, to understand the technology and terminology involved, to recognise the various benefits and pitfalls, and to ponder upon the creative opportunities that this changing paradigm presents.

The most pressing question on the mind of most readers right now is probably "How will loudness normalisation affect the sound of my mixes?" The answer is that, unlike Optimod-style broadcast processors, it won't change the sound you create at all; it will only change the volume. Note, however, that this change in volume can dramatically affect your perception of a track! In a peak-normalisation domain, the perceived loudness of a hyper-compressed track counteracts any sense of dynamic constriction. But when that loudness is taken out of the equation in a loudness-normalised environment, the dynamic constriction often becomes glaringly obvious and undesirable.

I mentioned earlier how loudness-normalised systems typically employ a relatively low loudness target level and have a relatively large headroom margin. That headroom margin is there to accommodate dynamic material, and it can be used to creative advantage. While the average energy level of an entire track determines its loudness, it's quite permissible to employ a large peak-to-average ratio to introduce dynamics and transients without affecting the overall loudness. Consequently, tracks with large crest ratios will inherently sound far more punchy, dynamic, and musically and aurally interesting than heavily squashed tracks. The differences aren't usually subtle, either! Loudness normalisation really punishes hyper-compressed music, revealing it to be the solid 'audio brick' that it so often is, while music with natural dynamics and transients shines like a beacon by comparison.

The two diagrams here hopefully illustrate these points better than my words. These are screen grabs from Cockos Reaper comparing two stereo tracks ripped directly from commercial CDs. The one on the left (coloured here in red) is Chris de Burgh's 'Devil's Eye', taken from an early 1980s CD release, while the one on the right (in green) is Grand Prix's 'Samurai', taken from a much more recent, remastered 'greatest hits' album. I apologise for my (lack of) musical taste, but these tracks highlight the issues nicely.

As you can see, the Chris de Burgh track isn't heavily limited or compressed at all, and it exhibits quite wide-ranging musical dynamics building and falling throughout the track. Using a BS.1770-compatible loudness meter I measured the True Peak level at -1.4dBTP, and the Integrated Loudness measured -18.8LUFS for the entire track. In contrast, the Grand Prix track has a true peak level of +0.5dBTP (in other words, some inter-sample clipping is present!) and an Integrated Loudness of -7.8LUFS, which is 11dB louder than the de Burgh track. This loudness disparity is painfully obvious if you play the two tracks back-to-back. Interestingly, I also have the original CD album containing that Grand Prix track (I was a closet head-banger back in the day!) and that version measures 0dBTP and -9.6LUFS. Clearly, the 'greatest hits' re-release suffered some further destructive compression and peak limiting during its re-mastering!

The lower screen grab shows what happens to these same two tracks in a loudness-normalised environment. In this case I employed a target loudness of -23LUFS - the HDTV standard, and a little lower than the current iTunes Radio setting. Now, the incredibly loud Grand Prix track (in green) has been attenuated to bring its loudness in line with the target value, with the result that its peak level is now just -13.7dBTP, with virtually no variation other than for the intro and a break towards the end. The Chris de Burgh track needed less attenuation to match the loudness target and, importantly, its more dynamic nature means its transient peak level reaches -4.5dBTP, a full 9dB higher than the Grand Prix track, despite both having equal overall loudness.

So, the track that was remastered to sound amazingly loud and 'in your face' has actually come out sounding flat, overly compressed and aurally uninteresting compared with the far more dynamic, but originally 11LU quieter, track. Not surprisingly, it's the Chris de Burgh track which comes across as the more interesting and musical, and is a more pleasurable listening experience. I'd encourage anyone to try this sort of comparison for themselves: hearing is believing.

As you can probably imagine, during the music industry's transition from a peak-normalisation workflow to one driven by loudness-normalised consumer outlets, there will be conflicting mastering requirements. Should a track be mastered to sound as loud as last year's music when played on CD? Or should it be mixed and mastered to be a little more dynamic so that it fares better when played on loudness-normalised radio and streaming services? Decisions, decisions!

It's not uncommon at the moment for chart artists (or their labels) to release loud CD and download masters for the general public, along with separate, less squashed, radio-play versions for the broadcasters that will better survive current radio processing technologies. Where the material is released on vinyl as well, another even less compressed master is often produced too, simply because vinyl can't cope with a lot of heavy compression and limiting.

I suspect a similar multi-release arrangement will permeate the mastering industry for a few years to come, as loudness normalisation gradually becomes de rigueur, perhaps with separate mastered versions optimised for loudness-normalised and peak-normalised destinations. I can also foresee some artists and labels revisiting their back-catalogue material and having it remastered to sound better in loudness-normalised environments. I can't wait!

As far as mixing is concerned, there's no need to change anything about the way you approach or balance your mixes; all of your existing skills still apply. The mix is whatever the music calls for, just as it has always been, whether that means dynamically varied or constrained. Some musical genres involve building the mix from the ground up deliberately to sound dense and powerful, and if that is what is appropriate to the work in hand then carry on in the same way. However, mixing a track to sound loud, purely for the sake of competing in the 'loudness wars', will clearly be a futile effort if the track is destined for loudness-normalised (ie. most) outlets. The practice of emphasising the mid-range energy of a mix, or curtailing the low end, which is commonly employed to maximise the perceived loudness, will be counter-productive.

Since loudness-normalised replay systems enjoy a reasonable headroom margin, the opportunity is provided to make use of that by allowing occasional loud transients, build-ups and break-downs. All of our senses respond to change, so transients are perceived as exciting and interesting. Consequently, mixes will have more impact and interest if you deliberately make use of transients such as emphatic drum breaks and general dynamic changes. I'll stress what I said earlier: the headroom is there to be employed to creative advantage! Just remember that loudness is determined by the average level - so occasional loud bits are fine. Continuous loud bits will give a higher Integrated Loudness value.

Referencing is the one area where some changes to working practices will be required. Ideally, reference tracks should be assessed for loudness and then auditioned at the same Target Loudness level as the mix-in-progress. That way, you won't feel obliged to compete in the loudness war, and you can concentrate on matching tonality and character. Of course, most back-catalogue reference tracks will have been produced to sound loud and so will be dynamically fairly flat. In comparison with that kind of reference a dynamic mix-in-progress will inevitably sound more sonically interesting. Most mix engineers and consumers will find that very refreshing!

To work in a loudness-normalised environment, I would urge the use of a BS-1770 compatible meter plug-in inserted in the stereo mix channel, and set up with an appropriate target level. In the absence of a recognised music industry standard at the moment, I suggest that the iTunes Sound Check level of around -16LUFS would be appropriate. The absolute target level is not particularly critical - it's matched loudness with your reference material that's the aim here, along with a reasonable headroom margin to accommodate wanted transients and dynamics.

The aim while mixing is to achieve an Integrated Loudness value for the entire track which is as close as possible to that target value, with the True Peak level not exceeding -1dBTP. If you already track and mix with a sensible headroom margin - as you should - this shouldn't be too different from your normal practice.

The difficulty most have when first starting to work with a loudness meter is in trying to hit the target level straight away. This is not necessary: the only loudness value that matters is the one showing when you get to the very end of the track. Most songs build gradually to a crescendo at the end, so initially you would expect the Integrated Loudness value to start low and gradually creep up as the mix progresses. At the end of the day, the meter is measuring loudness in a very similar way to our own perception, so if it sounds dynamically balanced to your ears the meter generally agrees! Also, don't obsess about hitting the target level precisely - that's the role of the consumer's replay system - just try to get it within a couple of LU.

The biggest workflow difference is what happens after you profess yourself happy with the final mix. Currently the next stage for most people would be a 'mastering pass' where the headroom is stripped out and the track made to sound as loud as the reference mixes. Clearly, that competitive loudness processing will no longer be necessary or appropriate - at least not as far as radio, streaming and download consumers are concerned. Instead, the mastering process can focus on optimising the spectral balance and achieving a consistent sound character across related tracks - or to put it another way, genuine polish rather than heavy sanding!

There's no doubt that large parts of the music industry are embracing loudness normalisation, and although it will take time to become the standard way of working everywhere, it looks certain that this change will become universal. In a loudness-normalised environment the current style of peak-normalised material doesn't fare well at all: the perceived loudness benefit is lost, and the sonically and musically destructive consequences of hyper-compression are clearly exposed.

Instead, loudness normalisation positively encourages the use of dynamics and transients. Tracks are made punchy by being dynamic rather than just loud, and compression and limiting become musical effects rather than essential competitive processing. Headroom is restored, inter-sample clipping is banished, and the digital environment finally achieves the sonic quality and dynamic range of which it is capable.

No-one likes change, and long-held working practices will take time to overcome, but loudness normalisation is win-win both from the perspectives of musical and sonic quality. There will inevitably be some short-term practical difficulties with monitoring levels in some consumer personal music players, and until all the DAW manufacturers incorporate BS.1770 compliant meters, some third-party plug-ins may be required. Overall, though, the benefits far outweigh the difficulties, and I urge you to familiarise yourself with the new metering and the creative opportunities that Loudness Normalisation presents.

In July 2006, the ITU published a paper with the catalogue number ITU-R BS.1770. The '-R' indicates that it is a 'recommendation' paper, and the 'BS' prefix means it falls within its Broadcast service (Sound) series. The code numbers don't mean much to anyone, of course, but the paper's title is considerably more informative: 'Algorithms To Measure Audio Programme Loudness And True Peak Audio Level'.

The original document has been superseded three times so far, with refinements being determined by the PLOWD group, which comprises 240 creative and technical expert members, including independent practitioners and members of a number of professional bodies and manufacturers. BS.1770-1 was released in September 2007, BS.1770-2 in March 2011, and the current edition BS.1770-3 in August 2012. All four versions are available from: http://www.itu.int/rec/R-REC-BS.1770 if you're interested in the evolution, but only BS.1770-3 is relevant today. The various updates reflect small evolutions of the loudness algorithm at the heart of the recommendation, mainly after feedback from the EBU and other interested parties. The current version of the algorithm is more sensitive to isolated loud peaks, and places less weight on low-level background sounds compared with the original. A 'loudness range' calculation to indicate the perceived dynamic range was also incorporated. Obviously, this means that any attempts to fool the system with clever mix trickery can be accounted for in future updates.

The bulk of BS1770-3 is concerned with defining a reliable, objective measurement of the subjective loudness of a piece of audio material, such as a TV programme, film, or advertisement. The aim is to allow material from disparate sources to be compared impartially (and automatically, if required), so that the subjective loudness can be matched between them with a high degree of consistency. The recommendation also details, in an appendix, the requirement and methodology for True Peak metering in the digital domain, specifically to avoid missing inter-sample peaks and thus inadvertent clipping in subsequent signal processing or format conversion.

The loudness-metering algorithm detailed in the latest BS.1770-3 (and in EBU Tech Doc 3341) document involves four distinct stages: response filtering, average-power calculation, channel weighting and summation.

The response filtering stage is called 'K-weighting' and it combines a boosting high-shelf equaliser with a high-pass filter. The shelf equaliser provides 4dB boost above about 2kHz and is intended to replicate the acoustic effects of the head. The high-pass filter simulates our reduced sensitivity to low frequencies when it comes to assessing loudness, with a 12dB/octave filter turning over at roughly 100Hz.

After this simple filtering, the next stage determines the average signal power using a mean-square calculation (ie. the average of the squared values of sample amplitudes). The power is averaged over a 400ms measuring period which is updated every 100ms, and the result may then be adjusted or 'weighted' before being summed with any other related audio channels. In a stereo system, the average power levels for the left and right channels are simply added together. In a surround system, the front three channels are added together, and the two rear channels are raised in level by 1.5dB relative to the front channels before they're added to the combination. This makes the metering slightly more sensitive to the rear-channel energy, which correlates more closely to human perception. In surround systems the LFE (low-frequency effects) contribution is ignored completely as far as loudness assessments are concerned.

Following this channel summation, the loudness value is given as a logarithmically scaled number in Loudness Units relative to digital full scale - so it will always be a negative number. The EBU prefer to use the absolute units of LUFS, while the ATSC in America use LKFS (loudness, K-weighted, relative to full scale), but the calculations are identical and the two terms are really interchangeable.

Where a relative (rather than absolute) loudness scale is more appropriate or convenient, the deviation in loudness from a given Target Loudness level can be given in Loudness Units or LU. For example, if the target level is -23LUFS but the material actually measures -20LUFS, its loudness offset would be described as +3LU.

Normally, the loudness value of any pre-recorded material can be measured and the replay level adjusted to ensure precise conformation with the Target Loudness, but where exact loudness normalisation is not practical - such as for live broadcasts - the current EBU R128 HDTV broadcast standard calls for a loudness deviation of no more than ±1LU.

Gating & LRA: Although the basic principles of the loudness algorithm are fairly straightforward, there's some added complexity to make sure that quiet periods don't drag the average loudness value down unfairly. For a start, if the signal level lies below -70LUFS it plays no part in the loudness measurement process. This makes sure that silent sections at the start and end of a programme don't affect the loudness value.

Secondly, if the audio level falls 10dB below the current programme loudness value it's also ignored, so that pauses in dialogue, or quiet sections between action sequences, are effectively ignored. In this way the loudness algorithm focuses only on the 'foreground' sounds.

Another facet of the loudness algorithm is the calculation of a 'Loudness Range' or LRA figure, which essentially indicates by how much the programme dynamics are being controlled. The LRA value is derived by analysing a continuously sliding three-second loudness assessment window, with any values more than 20LU below the current programme loudness level gated out. Again, this ensures that the LRA figure reflects the range of foreground audio signals, and ignores low-level background noise and the system noise floor.

The distribution of these short-term loudness levels is further quantified by setting 10 and 95 percentile limits so that the LRA figure is actually the difference between the loudness levels at the 10 and 95 percent points in the total measured range. The lower 10 percent limit prevents, for example, the fade-out of a music track from dominating LRA value, while the upper 95 percent limit ensures that a single, unusually loud, sound, such as a gunshot, cannot on its own be responsible for a large LRA value. A typical action feature film might have an LRA of 25, while most normal TV programmes would be closer to 10, and compressed pop music could have an LRA value of 3 or even less.

A key element of the BS.1770 recommendation is a True Peak meter to evaluate the actual peak level of the reconstructed audio waveform. This is important because, although the replay level of material is determined by its loudness, and a generous headroom margin is usually allowed, the physical medium still has a finite clipping level which must be protected

Critically, the True Peak meter algorithm recognises inter-sample peaks which are often missed by the simple sample-peak meters typically provided in DAW software and digital hardware. Sample-peak meters display only the highest amplitude value of a group of consecutive samples, and the scale is calibrated in dBFS (decibels below full scale), with 0dBFS at the top. To differentiate a True Peak meter reading from a conventional sample-peak meter, the new scale is marked in dBTP (decibels, true peak), and typically goes up to +3 or +6dBTP. The simpler Loudness meter plug-ins often don't show a True Peak meter at all, instead providing just an overload light which illuminates when the signal exceeds a defined threshold - usually -1dBTP.

To detect inter-sample peaks a True Peak meter employs a relatively high oversampling ratio which allows it to more fully reconstruct the continuous audio waveform. BS.1770 calls for a minimum oversampling ratio of 4x, which gives a maximum possible under-read error of slightly less than 0.7dB. For that reason, the EBU recommends setting the permitted maximum level (PML) to -1dBTP, thus guaranteeing that the signal can never cause an overload or clipping in any following digital processing equipment or converters. (The ATSC A/85 recommendation sets a slightly more conservative PML of -2dBTP.)

Not surprisingly, the higher the oversampling ratio, the smaller the possible inter-sample under-read. An 8x oversampling ratio gives a maximum error of slightly less than 0.2dB, and 16x oversampling is accurate to within 0.05dB. However, higher oversampling ratios involve a greater processing overhead, and in practical terms the benefit is negligible in most cases. So the BS.1770 recommendation is for a PML of -1dBTP based on the more manageable 4x ratio, which is a very pragmatic solution adopted by the vast majority of loudness meter plug-in manufacturers.

We are accustomed to working with level meters that share very recognisable display characteristics. For example, VU meters are instantly familiar, with their typically yellow background scaled with 100 percent modulation under 0VU, with the red section of the scale stopping usually at +3dB. Quasi-peak meters are also quite recognisable, with their typically black background and white markings - especially the BBC's PPM, scaled simply from 1 to 7. Similarly, all digital sample meters are bar-graphs, and scaled with 0dBFS at the top, going down to -50dBFS or more, and often with an expanded range over the top 6dB or so.

However, none of the loudness meter recommendation documents or regional implementation standards dictate what a loudness meter should look like - they only specify how the numbers should be calculated! I personally think this is a major failing of the standards bodies, because the natural consequence is that every single manufacturer that produces a BS.1770-compliant loudness meter does so with its own unique layout and graphics. Not surprisingly, the resulting variation confuses the heck out of most users, and acts to discourage the uptake of loudness metering because no-one is quite sure which version to buy! The good news, though, is that all loudness meters calculate the displayed numbers in exactly the same way, and they all show the same things, just in different places and in different ways.

The single most important parameter is the 'Programme Loudness' or 'Integrated Loudness' (I) value, which is the overall loudness of the entire programme item. This parameter is usually emphasised in the meter display somehow, often by being larger than any other number or in a prime location. The Integrated Loudness value should equal the Target Loudness value by the end of the programme.

The Integrated Loudness value is always associated with a start/stop/reset timer function, which determines the duration of the measurement. The system should be reset and started before the start of the programme material, and stopped at the end, either manually or automatically by being linked to the DAW transport commands in some systems. The actual pre-roll start and post-roll stop times before and after the programme aren't critical because of the -70LUFS gate which ensures the programme loudness value isn't affected by silences.

In addition to the Integrated Loudness value, most loudness meters display five other parameters one way or another - most commonly as simple numeric values in boxes somewhere, but sometimes with bar-graphs. The first two are usually labelled M and S. The M value is 'Momentary' and based on a sliding 400ms window updated every 100ms, so that it gives an indication of the instantaneous loudness. This is intended to aid initial level-setting when starting a mix, and in many meters the M value is displayed as a bar-graph, where it behaves in a very similar way to a standard VU meter.

The S value is the 'Short-Term' or 'Sliding' loudness value, which is based on a sliding three-second window. When mixing material on the fly, the S value is the one to keep an eye on as it responds reasonably quickly to mixing adjustments and provides a good indication of where the mix loudness is in relation to the target value, moment to moment.

The True Peak Level may also be displayed, either on a bar-graph or as a numeric value, and the normal maximum permitted value is -1dBTP (-2dBTP for ATSC A/85 installations). Anything higher than that indicates the presence of inter-sample peaks, which will cause internal overloads and clipping in any signal digital processing that involves oversampling, and that includes all modern D-A converters, sample-rate converters, lossy data-reduction codecs such as MP3 and AAC, and most EQ and dynamics plug-ins. Some meters don't display the actual True Peak value, but illuminate a warning indicator if the signal exceeds a preset PML threshold.

The final parameter, which not all meters provide, is the loudness range or LRA value. Currently there is no prescribed acceptable range, but a typical action feature film might exhibit an LRA of around 25dB, while a typical TV programme might be about 10dB. The LRA of music varies widely depending on the style and genre, but I would suggest somewhere between 3 and 15dB would be acceptable for most pop music styles, and a little wider for classical and jazz, etc. To provide some points of reference, Chris de Burgh's 'Devil's Eye' used in the earlier illustration has a pretty big LRA of 14.4dB, while the main 'brick' body of the Grand Prix 'Samurai' track has an LRA of just 1.5dB.

Most loudness meter systems also include some form of 'history' chart. Often this is shown as a horizontally scrolling graph of loudness levels, sometimes overlaying the Integrated, Short-term and Momentary loudness values on the Target Loudness threshold. An alternative method, used by TC Electronic, is to display the history as a rotating 'radar display'.

Since no two manufacturers display the loudness and related data values in exactly the same way, the choice of meter comes down to entirely personal preferences of manufacturer and graphical presentation aesthetics. The only technical issues to consider are conformity with the latest ITU specifications (currently BS.1770-3) and any regional requirements such as EBU R-128, or ATSC A/85, the latter affecting the display units and any preset Target Loudness and True Peak limits.

If you wish to check the accuracy and conformity of your chosen loudness meter, Prism Sound and Qualis Audio have jointly created an excellent and comprehensive suite of test signals, which are available on both of their websites (see 'Further Reading' box).

Many commercial loudness meter plug-ins include a facility to assess the loudness of files in the background and faster than real time. It may also be possible to work in a batch mode to assess a large group of files.

Apple's Sound Check facility was introduced with iTunes 3, way back in 2002, and it's included in all current iOS devices - iPods, iPhones, iPads and so on. The idea behind Sound Check is essentially to avoid the annoying (and potentially even dangerous) volume changes that can occur if your playlist moves from, say, a quiet jazz ballad into a full-on screaming metal track! It works in a broadly similar way to the ITU-R's BS.1770 algorithm but - as is Apple's usual policy - precise details of the system's inner workings are a closely guarded secret. Although Apple's algorithm appears to be not quite as precise or accurate as the BS.1770 algorithm, it is certainly close enough to give very usable results and benefits.

When Sound Check is activated for the first time, via a tick box in the Playback section of the Preferences dialogue in iTunes, the entire iTunes library is scanned and the loudness of each individual song or audio track measured and logged. This can take quite a while, depending on the size of your music library - expect somewhere in the region of 50-100 tracks scanned per minute. Once scanned, the loudness data is stored in the 'normalisation information' tag of the ID3 metadata associated with the individual MP3, AAC, AIFF or ALAC (Apple Lossless) files. Loudness metadata can also be stored in the iTunes XML database for supported file formats that don't include compatible metadata.

When a track is subsequently replayed with the Sound Check function switched on (enabled in most iOS devices via the Settings menu), the stored normalisation information is used to offset the internal replay level for that specific track, such that it ends up at broadly the same perceived loudness as every other track. The system is easy to use, easy to set up, and is generally very effective, particularly for anyone who uses the 'shuffle mode' with an eclectic music library, or who creates varied playlists!
Album Mode
While Sound Check's loudness normalisation is ideal when listening to disparate tracks, to maintain reasonably consistent listening levels, it's not so good if you want to listen to a complete musical work where the relative loudness of sequential musical sections is an important factor in the overall experience. This applies to any musical genre where deliberate contrasts of quiet segments between louder sections are aesthetically important - such as in complete classical symphonies, concept prog-rock albums, or any albums containing a diverse selection of material.

Up until recently, if Sound Check was switched on but you chose to listen to a complete album, any intended loudness variances between tracks would be removed, destroying the intended effect and much of the musical enjoyment! Happily, Apple have now corrected this weakness: the latest version of iTunes now has an unannounced but very welcome Album Mode which intelligently applies loudness correction depending on how you have chosen to listen to your music.

If Sound Check is turned on with tracks played with the iTunes shuffle mode, via a playlist, or through the iTunes search screens for artist or genre and so on, loudness normalisation is applied and the relative loudness of each successive track remains pretty consistent. In contrast, if music is selected through the iTunes album listings, the player assumes you want to enjoy the tracks with their original relative levels, and so the Sound Check mode is temporarily modified. This is a useful step forward, but unfortunately it also means that switching between album selection and artist or track selection results in quite a major listening level shift because the latter is loudness normalised while the former isn't. So this isn't quite a perfect solution yet, but it's much better than the original options, and for the first time it is now quite acceptable to leave Sound Check turned on. In fact there are industry calls for Apple to activate Sound Check by default in future software updates.
Not Loud Enough!
A problem which is a little harder to resolve is that of inadequate playback volume when Sound Check is enabled. Clearly, for loudness normalisation to be effective, the Target Loudness level has to be low enough to accommodate the peaks of the most dynamic material likely to be encountered. Apple appear to have chosen a target level of around -16LUFS, and this inevitably means that the playback volume for highly compressed material will be considerably lower than its native level when auditioned with Sound Check switched off - in practice, as much as 12dB lower, which represents quite considerable attenuation.

The obvious answer is simply to increase the device's playback volume control to restore a comfortable listening level when Sound Check is enabled. Unfortunately, though, this isn't always possible, particularly for Apple device users in Europe where 'hearing safety' concerns led Apple and other manufacturers to restrict the maximum playback volume in iPods, iPhones, iPads and even laptop computers! Hopefully future firmware updates will enable increased playback gain when the Sound Check function is active, but until then the only pragmatic solutions are to invest in much more sensitive headphones/earphones, or to employ an external headphone amplifier of some kind. Both involve additional expense and will therefore dissuade many consumers from using the Sound Check facility, at least in the short term, which is regrettable.

Crest Factor: A measure of a signal waveform amplitude, indicating the ratio of the peak to average value. A crest factor of 1 indicates no peaks (ie. a DC signal), while a sine wave has a crest factor of 1.414 and a triangle wave 1.732.

dBTP: True peak level in decibels, with reference to digital full scale. The maximum value of the reconstructed audio signal waveform, recognising any inter-sample peaks.

I, or Integrated Loudness. The loudness value indicated on a loudness meter, which is integrated across the entire programme duration. Also known as Programme Loudness. This value should equal the Target Loudness value by the end of the item.

LKFS: Loudness, K-weighted, with reference to digital full scale. ATSC preferred term for the absolute loudness value. (See LUFS).

Loudness Normalisation: A working practice in which an arbitrary loudness level defines the reference point, and the loudness of individual signals is assessed so that they can then be adjusted to match loudness on replay. This approach imposes a headroom margin and thus encourages dynamic variations. (See also Peak Normalisation).

LRA: Loudness Range. The distribution of loudness within a programme (an indicator of the programme's perceived dynamic range).

LU: Loudness Units. A relative loudness value from the Target Loudness level.

LUFS: Loudness Units with reference to digital full scale. The EBU preferred term for the absolute loudness value. (See LKFS)

M, or Momentary Loudness: An instantaneous loudness value shown on a loudness meter. Assessed from a 400ms time window updated every 100ms. Used only for initial level setting.

Peak Normalisation: A practice whereby the highest peaks of different programmes are aligned to a defined reference level, which encourages engineers to use dynamic range compression to increase perceived loudness.

Programme: An individual audio programme, advert, music track, and so on.

Programme Loudness: The Integrated Loudness over the duration of the programme item, in LUFS. Also shown as the 'I' value on loudness meters.

Target Loudness Level: The intended loudness-normalisation level.

TP, or True Peak: The maximum peak value of the reconstructed audio waveform (which includes any inter-sample peaks).

Sample Peak: The typical digital sample-peak meter system which indicates the highest amplitude value represented within a batch of digital audio samples. It is typically lower than the True Peak value of the reconstructed analogue waveform, because it ignores any inter-sample peaks.

S, or Sliding or Short-term Loudness: A short-term loudness value shown on a loudness meter, assessed from a continuously updating three-second time window. Used for moment-by-moment mix adjustments.

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As the nails are being hammered firmly into the coffin of competitive loudness processing, we consider the implications for those who make, mix and master music. In a surprising announcement made at last Autumn's AES convention in New York, the well-known American mastering engineer Bob Katz declared in a press release that "The loudness wars are over."

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Did you combine the racks or some modules for Sunergy?
KS: Yes, it was a little bit of both. It was both of us in our own set-ups and then we would take turns clocking each other and then we would feed of each other.
SC: We could integrate the two systems rhythmically. Also, we had a set of shared raw material. So each chose the sets of pitches that we would use. And we are both pitch-oriented composers and electronic performers. Some electronic music is more based on abstraction, or noise or whatever. We as composers love using pitches because we both have a classical root system. It was a lot of fun because it was in the moment and I think it really was synergetic. We complimented each other a lot of times. Kaytlin always contributed something what was needed, missing or useful. Maybe it was a bass or an improvised line above.

Was there any remarkable accident that occured which make you find something that you have not found before while you recorded together?
KS: It feel like it was all happy accidents! We kind of had, for the lack of language, like a jazz-mentality about it because we were improvising the whole time. It was all happy accidents.
SC: For my Buchla concerts from 1975 that just came out on Finders Keepers, I had written a documentation about how to perform the Buchla which was included in the collectors' version. I referred back to that cookbook and took this a starting point. If you listen to the 1975 concerts, it would not sound like anything Kaytlin and I did together because it was a different instrument and a different time, it was a 200, and not a 200E. There was no memory in the 200 but there was more controls. The thing about the Buchla, you know, the fascination, it was never the sound. The sound of the Moog is even is richer than the Buchla, but it was the way you could move the sound. It had a spatial controller, it had a lot of controlled voltage. It had a thing that was like a 3-D sequencer, in the MARF, the multiple arbitrary function generator. These things were complex. And the new instrument is not even as complex as the earlier one but it does have a memory.

Video: Kaitlyn Aurelia Smith & Suzanne Ciani - Closed Circuit

Speaking more metaphorically, what is then electricity to you while you are making sound with the Buchla or another synthesizer?
KS: That is one of my favourite things to think about when I am making music. When I first learned it, that was on a 100, which I am sure Suzanne has also played.
SC: Yes.
KS: At least with the one I was playing, the impedance level was just not really haunting in at this time. You could really feel these pulses of electricity. If I turned on the light switch, you could hear the difference and depending on what monitors I was using, I could hear the pumping of electricity. It feels like this live force and that is something that I first started to relate to and I come back to that thought a lot. It sounds abstract but it is my personal relationship with it.
SC: Also, for me the underlying electricity is an actual source of stability. The consistent cycling, the perfect rhythmic pulsing is something that I think has a subliminal comfort and allows us to feel secure and relaxed.
KS: Oh, that is so interesting. I like that.

That is a good transition to electronic club music. Techno has this steady pumping and pulsing beat.
SC: Yes, that is all wonderful. And yes, I think Techno is based on that pumping stability. But for me it is just too one-dimensional because as a composer I just like more... I don't want to say complex, it is not about complexity. It is about a menu. If you go for dinner, you don't want to eat just carrots.
KS: That makes me think about when you were first asking about why modulars. I grew up playing African music and so poly-rhythms have always been where my foundation is, rhythmically. Modulars really help you play those as one person, because usually you would need multiple people to play that.

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Kaitlyn Aurelia Smith & Suzanne Ciani: Constantly Working

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Did you combine the racks or some modules for Sunergy? KS: Yes, it was a little bit of both. It was both of us in our own set-ups and then we would take turns clocking each other and then we would feed of each other. SC: We could integrate the two systems rhythmically.

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Described by Pitchfork as "one of the most innovative voices in British Music" Anna Meredith is a genre-crossing composer and producer whose work straddles the worlds of contemporary classical, art pop, electronica and experimental rock.

We're delighted that she is on board as an ambassador for our Both Sides Now programme and we're lucky to have worked with her on two exciting residencies over the past couple of years. Most recently Anna led a group of women to create new music and visuals as part of the final instalment of The Hexagon Experiment - an exciting six-part series bringing together pioneering women at the forefront of music, art and science.
Anna has shared with us some top tips for aspiring musicians to give you an insight into her creative process. Give them a read...
1. Denial. Don't get too caught up in the pressures of a project - who's going to listen to it, be marking it or there at a gig. I find all those worries don't help you focus on what you think makes the music work. Write on your own terms.

2. Social Media. We all know it (though easier said than done) but getting too fixated or caught in the endless stream of Twitter/Facebook/Instagram or looking for a little compliment boost on something you post can be so destructive. While it's of course important to keep people up to date with things that are happening, try to post stuff then leave it alone.

3. Learn Tech On Your Own Terms. It's very easy to feel daunted by equipment or software you don't feel you totally understand and say "I'm not good with technology"and leave it there. Find a really simple way in that works to your strengths and leave it there! Sure, the programme might be able to do 1000 other thing but if you just want it to be say, a looper or an echo or for simple edits that's great! For example, I use Sibelius, the classical notation programme to write my electronic music before going to specialist programmes like Ableton, only because that's where my skills lie.

4. Find Your Friends. Having the support of other people who are also writing and experimenting is invaluable! Not only do you have immediate access to people to ask questions about their areas of expertise or try out ideas, but I find I feel more confident when I can see my friends writing interesting music and doing things on their own terms. I still have a bunch of friends from college who are the people I sent bits of music too when I'm stuck on what to do next.

5. Have A Plan. When writing music I'm not one of those people who can improvise their way through a piece. I like to have a plan for the shape of the material. I do this by drawing a timeline and drawing graphic shapes to show the drama or story telling of the material I'm working with. Then I'm sort of 'auditioning' my own ideas to fulfil the kind of dramatic shape I need them to be. Being able to stick this plan up on the wall and refer back to it while I'm writing means I don't get totally lost.

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My top tips with Anna Meredith - Brighter Sound

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Described by Pitchfork as "one of the most innovative voices in British Music" Anna Meredith is a genre-crossing composer and producer whose work straddles the worlds of contemporary classical, art pop, electronica and experimental rock.

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Influenced by UK club music from her time spent working in record stores, the ambitious DJ/producer has a history of merging the organic and synthetic, creating rematerialised compositions for a number of sound art commissions.

"Once you understand how sampling works you think, hmm, maybe I don't need to play an instrument. It opens you up to other things."

The former NTS Radio DJ is currently resident at Somerset House Studios, where her small but focused studio belies the complexity of her solo debut LP, Workaround. A contender for experimental electronic album of the year, it shows Dillon's inventive fusion of acoustic samples and groove-driven modular permutations - a breath of fresh air for the UK electronic scene.
Has your background in the visual arts informed your approach to music making?
"In terms of having a curiosity about art, sound design and film, it definitely all feeds in. I once mentioned that attending art school was a waste of time, but I didn't mean that to be disrespectful to the college. It's just that most people think that if you go to art school you'll spend all day life drawing but, because art isn't rooted in the traditional methods it was, it teaches you more about thinking, history and concepts than techniques. I really enjoyed that, but I wonder if that was a good use of my student loan."
You worked as a BBC Radio 3 researcher...
"I started off doing internships at different radio programmes and worked for a production company that made jazz and world music for Radio 3. It was unpaid, but if you stuck around long enough jobs became available."
Sampling appears to be a strong tenet of your music-making discipline...
"I remember saving up to buy a Yamaha SU10 sampler when I was a teenager and plugging it into the record player. I didn't really know what it was and couldn't imagine there was a whole load of people doing this for a living - I just thought it was a good way to loop favourite bits of songs together."

(Image credit: Future)

Something must have led your curiosity towards buying a sampler?
"It was quite an isolated decision and a bit of an anomaly. I probably didn't even know what music software was, but once you understand how sampling works you think, hmm, maybe I don't need to play an instrument. It opens you up to other things.

I was mad about the DJ Shadow album Endtroducing..... and thought it was the most incredible collage of sound. I had a grip of it coming from someone making a record from a record collection as opposed to a group of people in a room playing as a band."
Your album Workaround is very progressive. It sounds like you have a novel approach to sound design?
"I tend not to learn how to use gear methodically; I'm actually a bit impatient. I learn gear up to a point, get bored and use something else, but I found learning how to use a computer to make music really liberating because I didn't know what I was doing.

"From that point of view, I feel that not knowing what you're doing is better than mastering a piece of gear. There's this weird rumour about me that I've had classical training, but I really haven't. You can get programmed by technique and lose yourself. In terms of sounding progressive, labels like Hyperdub are really future-oriented, but I don't have that urge to create something that sounds brand new, it's just got to sound new to me."
Are there artists operating in the realm of sound design who influenced you?
"I'm really interested in Mark Fell as a producer, artist and thinker. He has a really interesting perspective in that he makes music that sounds interesting to him and doesn't care whether it's strictly suited to a club or an art gallery. Although he's clearly listened to a lot of club music and knows how those structures work to keep people dancing, he has his own take that I find endlessly inspiring."
Not making music for a specific category must be liberating but quite a brave step to take?
"It is quite brave I suppose, or naïve because you've got to try and make a living [laughs]. It's not about making something weird for the sake of it or a big statement about producing experimental music. It would be a bit disingenuous if I started making a certain type of techno music, for example, because I don't know about that style of music in any real depth. It's just about trying to find what the sum of all your weird influences is."
What environment best suits Workaround?
"If it fits anywhere it's probably club music. There have always been producers in club music doing their own weird thing - taking what they like about that scene and adopting a minimal or experimental approach. Over the last ten years in the UK and Europe, a lot of artists have been playing with those formulas and doing weird things with kick drums or timing.

"I can find a lot of what I'm interested in in Lee Gamble's music. He's an original Junglist, but makes a weird version of it."
The album has a certain consistency, not just in terms of the vibe but the sounds used. Did you deliberately restrict your sound palette?
"Yes, definitely. When you switch on your computer, for the first ten minutes you're buzzing and then you realise there are too many options and hit a wall. It's crazy that you can do anything, and that's what gives you the headache. I'll usually say, three colours, go... any constraint to get me going. Then you get really attached to those few elements and look to do more with them rather than piling on elements. I like people who do a lot with a few elements."

"When you switch on your computer, for the first ten minutes you're buzzing and then you realise there are too many options and hit a wall."

Does that primarily apply to working in the box?
"I use Logic and have set templates with a few sample banks that I know really well from things I've made in the past. Logic's always been my choice of DAW, mainly because it's the first one I used.

"I use Ableton for live and wish I was better on it because I can see how it's much more flexible, especially for making arrangement decisions."
The music sounds very fluid despite the grid...
"I sometimes wish I didn't use Logic simply because it is quite gridded [we spoke to Beatrice before the release of Logic Pro X 10.5]. I'd love to get onto Max/MSP because it's non-linear and gets you out of that thing where you're conscious of being a certain amount of bars into a track. It makes you listen to music in a different way. I've been trying to learn it, but I don't have the weeks and hours needed at the moment."
In order to move forward, is it essential to learn new technologies?
"Well, saying you can't do something can also be quite a good distraction technique - you can make something amazing with one drum machine if you have the discipline.

"There are so many examples of people being creative with electronic limitations. Lee Perry is an obvious example. All he needs is a spring reverb, a vocal and a drum machine to make some of the best music you've ever heard."
Is that a good principle to abide by for anyone taking their first steps into production?
"It's definitely fun to buy gear and I don't pass judgement on anybody who wants to buy loads of stuff. If you've got the money, why not, but it's easy to think that's your concern. Your concern should be getting what you want to say out, not fiddling around looking at gear on eBay all day - as tempting as that is."
Indeed, the title Workaround seems related to self-imposed limitations?
"Yes it is - you're the first person to pick up on that. It's no more complicated than the process behind how I made the album."
The beats have a sort of Afro-Caribbean tinge and I wondered whether these were sampled from those types of instruments?
"A lot of it is based on samples coming from the Mutable Instruments module Clouds. I guess most people use it for granular synthesis, but somebody made some firmware that you can download and replay through it so you can basically turn it into a resonator.

"You send things in and out comes this strange, resonated re-pitched version. I realised I could send drum patterns into it and these melodies would come that I could then move about. When I worked out how to do that it was really exciting, so any sounds on the record that are not obvious drum samples are probably resonated, pitched samples from Clouds that I turned into percussion - making my own sample pack, in effect."

Y ou also invited a lot of collaborators to sample sounds from...

"There's a lot of collaborators on the record, but it's completely fictitious in terms of me not needing to be in the same room with a bunch of musicians. I basically made two demo tracks at 150bpm and invited people to play all sorts of different takes on those two tracks. When I got the material in from everybody, I got rid of the backing tracks and had loads of material that I could then cut up and make new music out of.

"At times, the record might sound like people in a room playing along to something, but they're not. Again, it goes back to that collage thing where I'm looking to rejig everything in a way that electronic music gives you that freedom to do."
Did you want those sounds to stay acoustic or develop them into something more interesting?
"Maybe, or maybe it's just to make weird connections with things, which could be a complete disaster and sound horrible or be quite interesting.

"I remember hearing a programme on Radio 2 when I was ten or 11 describing how a Motown hit was produced. I thought it was amazing to hear somebody talking through all of the elements. It filled my imagination because you can feel really empowered when somebody walks you through the construction of a track. I suppose I'm as interested in how records get made as making music."
The electronic sounds/samples are very delicately syncopated. Was that the case or does the music belie a more improvised approach to its construction?
"There is a lot of pushing blocks of audio around the screen. Sometimes I think it would be amazing to be able to play an instrument in real time and get lost in that, but I've learnt things the other way round. So yes, the music's quite meticulously arranged. Hopefully the jigsaw puzzle flows and it doesn't sound like a maths lesson [laughs]."
The melody comes from the percussion rather than traditional synth lines?
"My fixation is rhythm, basslines and the textures around those things. If I ever have any money I don't go out and buy a synth, I'll buy a drum machine and that informs the shape of the album.

"The live set is the record with all the acoustic bits stripped out and replaced by synths, so it's a completely different entity. It's like a compressed version of the record, with the rhythms pushed up front and made for a club sound system where you can get that fantastic range of frequencies."
Do you take your hardware out on the road?
"I don't use any of my studio gear live; I run it off my laptop and make arrangement and processing decisions. At some point I want to use some of my studio sequencers on stage, but I need to write the music first or at least get some ideas together for doing that."
You mentioned earlier that you fixed the tempo to 150bpm throughout the album...
"I did some tracks a few years ago at 150bpm out of complete frustration. If you're doing 4/4 music it's usually got a vibe and emotion to it, but if you're playing around with kick drums in weird places and don't have a solid beat I find the music can seem quite slow because there's nothing driving it forward. House music is generally 120-125bpm and jungle is around 170, but 150bpm leaves a lot of space for perceived slowness or fastness."
You have a modest modular setup...
"I find the whole modular synth thing a real turn off in terms of how excessive it can get and the idea that whoever has the most gear has better ideas - I don't think that's true. I bought two Grids modules and Clouds and they're both all over the album. People will argue against it, but it's nothing you can't do inside the computer - I just found it really nice to get hands on and away from the screen. The Vermona module is a junction box/audio interface that allows me to sync the modular rack to the computer."
What do you get out of using Grids?
"It's a sequencer, but rather than the Roland TR-26 step sequencer I have, and all the cousins of that drum machine where you're always thinking about rhythm in 16-step sequences, this is a completely different approach.

"There's no linear sequencer laid out for you visually; it's just a bank of pre-recorded drum rhythms. You can swerve between one rhythm and another on the X and Y axis, which means you're shifting the bias, accent or swing of a rhythm.

"There are three ring dials, so if you imagine them as a kick drum, snare and hi-hat, you can basically dial in the density and send it to a drum machine. If you think of the harmonic variation quantities you have in the three dials, which go from 0 to 250bpm, you've got an incredible amount of intervals of rhythm, so it's brilliant to play with."
Grids is triggering your Nord Drum 2?
"The Grids sequencer sends a trigger out into the , which is a really nice digital FM synth for percussive sounds. It's got everything from incredibly icy, metallic sounds to 808-type beats. It's just a way of coming up with some interesting drum patterns, and within the Nord Drum there are endless parameters for shaping a sound. I think they'll both be in the live set eventually."

(Image credit: Future)

You also have the Zularic Repetitor, which is another rhythm-based module?
"That's another sequencer that works on a similar principle to Grids. There's a brain behind the panel stored up with loads of drum patterns. Again, the dials help you to swerve between patterns, but if Grids is loaded up with common western drum patterns, the Zularic Repetitor is pre-programmed with African and Indian polyrhythms.

"I'm not someone who understands the complexity behind these modules; I've just worked out my own way of using them. I'd love to do live sets with these because sequencing in real time is so exciting."
But a little scary I expect?
"Totally, because every day would be different and it's the opposite of the computer where everything is controllable. These modules take you on a wild ride and you don't know where you're going, which is probably why people are obsessed with them. I think you have to give them strict parameters for live work, and because of the way I make music it wouldn't represent the album because it wasn't made with improvisation in mind. That would have to be a separate project."
With the addition of the Vermona DRM-1 modular, you seem to have almost every drum tweaking option you could ever want?
"It's a mad one that. I actually bought it quite a long time ago for a sound art-type commission and used some of the fee to buy it. The Vermona's interesting because there's no in-built sequencer; it's just a sound source. I use it with Grids because it's got six trigger inputs or more and each row is an instrument. For example, the top row is the kick and you have all the parameters to shape that sound. There are so many variations that you could do a whole live set using just that drum machine."

(Image credit: Future)

Should we give more credit to these machines for directing our creativity?
"What's exciting about this technology is that while you half know what you're doing, the other half is the instrument shaping your ideas. Mark Fell talks a lot about people who say they have an idea, know exactly what they want and sit down and do it, but that's not really true because you'll have an idea about what something might sound like but you're navigating in collaboration with the machine."
You said that this music could all be done in the box, but presumably there's less post-production required using a modular setup?
"I think that's down to taste really. I love records and CDs and they both have their own quality, or maybe I don't know enough about sound to think they're different or one is better than the other. I'm sure there are people who really know what they're doing and can monitor the difference, and I'd trust their judgement over mine, but I'm not an audiophile.

"I care about the sound deeply, but I'll use what I can within the limitations I have. If you're going to listen to music on a laptop, what's the point? I find myself playing a Villalobos track on a laptop and thinking, this is stupid - I'm losing all of what he's got to say. It's questionable what people are going to get from your painstaking tweaking of sounds, but then you have people walking around with Beats by Dre headphones which are insanely bassy."

"Your concern should be getting what you want to say out, not fiddling around looking at gear on eBay all day"

Is your DAW complemented by any software plugins?
"Chromaphone is an interesting plugin; it's based on physical modelling synthesis. I use it to make drum sounds.

"If you stand at a piano and record every single angle you can build a comprehensive sample pack, but each sound is fixed even if you can shape it. The way I understand physical modelling synthesis is you can literally mimic the physical properties of a three-dimensional object.

"For example, we could take a glass and build it into an instrument by programming its dimensions and the material it's made from. It's like designing a sound-resonating chamber and the sound design options are infinite compared to a sample library where you're dealing with just one sound and only adding a filter or reverb. You could use Chromaphone to recreate the sound of a harp or a piano, but why not use the real thing? It's great for taking things to extremes and endless fun to use.

"Other than that, I use some Native Instruments software like and , and I've recently been looking into using and ."
You were quoted as saying "The computer always wins". What did you mean by that?
"When I knew that I wanted to make a record from electronic and acoustic instruments, I thought it might sound a bit cheesy. I didn't want to go off on some symphonic tangent - I wanted to make electronic music with guest components. I love how the computer can be quite dry, sober and rigid. The computer always wins was my way of saying to myself, don't go off on an acoustic tangent. The new material I'm working on is a completely different shape to Workaround - it's all solo stuff. No collaborations, just me and this room."

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Beatrice Dillon: "I feel that not knowing what you're doing is better than mastering a piece of gear"

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Influenced by UK club music from her time spent working in record stores, the ambitious DJ/producer has a history of merging the organic and synthetic, creating rematerialised compositions for a number of sound art commissions. The former NTS Radio DJ is currently resident at Somerset House Studios, where her small but focused studio belies the complexity of her solo debut LP, Workaround.