Q. Can you identify my discs?

Following a recent lesson I had taught on music technology history, one of our teaching assistants mentioned that she had found some five-inch discs at home in her loft.

 

David Cooke's mystery aluminium discs. 

They appear to be thin steel discs coated in what seems to be thick varnish. The groove is clear to see but the material has cracked in places. I tried to get my Technics deck to play them but they are too small and the arm returns too soon. I have a gramophone, but I think that the arm and needle could scratch the surface off.

Do you think that they could be some sort of home recording? Perhaps pre-tape, or at least pre-consumer tape machines? Possibly even 'end-of-the-pier' booth recordings?

David Cooke

Technical Editor Hugh Robjohns replies: I think you are right, David. These look as though they are the kind of discs that might come from an end-of-the-pier or music shop automatic booth recorder, dating back to the 1940s, at a guess.

It's all well before my time, of course! So, I made a few enquiries of my elders and betters (many thanks, Pete), and was told that five-inch discs such as these were made in automatic booth recorders. The discs play at 78 rpm and the recording was usually cut directly into bare aluminium. However, some discs were made with coated aluminium, as these pictures appear to show. The coating was either acetate or a type of gelatine that was water-soluble (and easily damaged, as you can imagine). It's hard to tell from the photo which type these are, but the obvious shrinkage and peeling means that they need to be transferred fast if the material is of some value, as this deterioration will almost certainly accelerate.

Something else to bear in mind is that a lot of these records were made using the old 'hill and dale' groove technique. This technique used vertical groove modulation, rather than the side-side modulation format that became the norm for commercial discs (or the 45/45 format used in modern stereo microgroove records).
Again, it's hard to tell from the photo, but the grooves on the pictures make me think that these are hill and dale recordings (unless the light is playing tricks).

Published May 2008

Korg MS-20m Monophonic Synthesizer Kit and SQ-1 Step Sequencer

Solve Latency Problems

By Martin Walker

The focus this month is on checking and improving the performance of your PC, with a new benchmark test just released and news of a useful tool that could help you pin down the source of latency-related problems.

As the majority of hi-tech musicians these days run lots of plug-in effects, how many simultaneous plug-ins a particular PC can run is a good indication of its limitations. A 'benchmark' test such as DAWbench (www.dawbench.com), which started life as a real-world song and then ignored the application's 'CPU meter' in favour of adding more and more plug-ins across 40 audio tracks, until audio glitching was heard, is therefore a good way to compare the relative performance of different PCs, or check whether your PC performance is all it should be.

While the original DAWbench Blofelds DSP40 test was for Cubase/Nuendo users only, this month sees the release of a new 'DAWbench DSP Universal' benchmark test that will be able to run on any multitrack audio application, and finally lets us compare the performance of different sequencers running the same plug-ins.

A Universal Benchmark?

The new test relied on finding a plug-in that was sufficiently taxing to fully load today's fastest PCs without requiring hundreds of instances to be launched, yet was freely available to all. Over the last six months or so, Vin Curigliano and other members of the DAWbench forum have evaluated various alternatives, including iZotope's freeware Vinyl and the 30-day demo version of Sonalksis' CQ1. However, the most suitable candidate proved to be ReaXcomp, the multi-band compressor bundled with Cockos' Reaper, which is now also available in a freeware VST plug-in version (www.reaper.fm/reaplugs).

 
The new DAWbench Universal benchmark not only lets you compare the audio performance of different PCs, but also of the same PC running different sequencer applications. These initial results are for three separate applications running on an Intel eight-core machine.As I write this, the DAWbench DSP Universal benchmark is already downloadable as a song template for Cubase, Nuendo 4.1.2, Reaper 2.1 and Sonar 7.02, with more applications to follow. It comprises four stereo audio music tracks, one stereo sine-wave audio track with no effects, so you can easily hear the first signs of audio glitching, plus 40 stereo sine-wave audio tracks, loaded with a variable number of plug-ins, which you activate one by one until you hear audio spikes or break-up.

Initial tests with quad-core and octo-core systems suggest that Reaper is consistently in the lead, letting you load every available CPU core to almost 100 percent (as shown in Windows Task Manager) before audio glitching occurs, whereas the other applications don't manage to balance the load as evenly between cores, and therefore can run fewer identical plug-ins before glitching kicks in.

The differences between Nuendo and Reaper increase as buffer size reduces, but can be up to 10 percent at 64 samples and below with a quad-core system, and 20 percent with an 'octo'. The current version of Sonar is lagging behind the other two applications at the moment.

While sequencer feature sets may differ hugely, for the many musicians who judge performance by how many plug-ins they can run on a particular PC these results will be a revelation. Now that we can directly compare the performance limits of different applications running the same audio tracks and plug-ins it will be interesting to see how users react.

BIOS Updates Can Improve Audio Performance!

In SOS January 2008 I discussed multi-core processors, and in particular the benefits they might provide with audio applications. I also mentioned that Cubase 4 and Nuendo 4 didn't currently provide all the benefits they could at low latency with a dual quad-core system, and that Steinberg developers had already acknowledged the problem and promised to address it in the first maintenance updates after the release of Nuendo 4 and Cubase 4.1.

Well, there's been an unexpected twist in this tale. Although plenty of musicians with cutting-edge octo-core systems experienced the low-latency limitations of both Cubase and Nuendo, one of the people who spent the most time and energy investigating these issues, using his DAWbench tests, was Vin Curigliano, who has also stumbled recently across a way to improve matters.

All he did was perform a routine BIOS update on his Dual Xeon quad-core 5355 PC (with eight cores running at 2.66GHz), and was flabbergasted to find he could run around 30 percent more plug-ins with a buffer size of 256 samples before his PC starting glitching, and a massive 130 percent more at 32 samples!
Although this BIOS update was specific to his particular motherboard, and the list of fixes in that update didn't mention anything that might suggest the source of these improvements, it's surmised that they are due to Intel 'microcode' updates — tweaks and bug-fixes for Intel chips that are installed by the BIOS (or the operating system) each time you boot your PC. Chip manufacturers sometimes send these out to motherboard manufacturers, and they then get discreetly bundled into the next BIOS update.

This supposition was strengthened when other users of octo-core systems featuring Intel's 5000X motherboard chipset (including Dell's Precision 690) discovered similar improvements after their own routine BIOS updates. Octo-core performance with other applications (including Sonar) also improved after these BIOS updates, but not by such a significant amount, perhaps due to the different ways in which each application splits the various audio tasks into multiple 'threads'. It will also be interesting to see if owners of eight-core Mac Pro computers featuring Intel Xeon 'Harpertown' processors experience similar improvements as a result of an EFI (Extensible Firmware Interface) update from Apple.

Spyware Doctor With Antivirus

So many musicians are now running audio software on PCs while connected to the Internet that I make no apology for returning to the subject of virus and spyware utilities and their impact on PC performance.

 
Why run separate spyware and anti-virus utilities on your PC for Internet protection, when you can run Spyware Doctor with Antivirus 5.5 as a single task, and potentially reduce operating system overheads?In PC Notes October 2007 I discussed ways to adjust virus checkers to run on demand, rather than constantly in the background, thus minimising their effects on real-time audio performance. In my experience, virus attacks are becoming far less frequent, so I wasn't too concerned about offering this advice, but other 'malware', including spyware, auto-diallers, keyloggers and trojans, is still on the increase, and its components can reside in many different areas of your PC. So, if you insist on connecting your music PC to the Internet, I would recommend that you always leave a background spyware checker running while you're connected, since it's generally far better to stop nasties getting into your system in the first place than attempt to remove malware that's already infested your PC.

In June 2007's column I reviewed the excellent Spyware Doctor version 5.0 from PC Tools (www.pctools.com). I mentioned in that review that you could buy it with an optional upgrade to integrate the functions of the PC Tools Antivirus utility, and also that combining spyware and virus searches was likely to speed up scan times and reduce the potential for conflicts, compared with using two separate utilities.
When Spyware Doctor was recently upgraded to version 5.5, I decided to check out this integrated approach myself and see whether I found running Spyware Doctor with AntiVirus 5.5 better than running separate spyware and virus-checking utilities. Version 5.5 incorporates further advances in removing 'rootkits' (stealth programs that run at a 'lower' level than the user can see with normal software utilities) and protecting network Registry settings, and can identify potentially malicious threats earlier, for enhanced real-time protection. It also provides a simpler categorisation of threats and has a smaller RAM footprint.

Doctor's Orders

After installing Spyware Doctor with AntiVirus and accepting its offer to activate various OnGuard real-time protection routines, I disabled my previous real-time virus-checking utility (Avira's freeware AntiVir PersonalEdition Classic, from www.free-av.com), and let SDAV carry on as the sole spyware and anti-virus monitor. During the following weeks it caught every nasty I exposed it to, blocking downloads from suspicious web sites, preventing Windows Explorer from opening infected files or emails, preventing malicious changes to my browser, and so on.

I experienced no noticeable extra overheads as a result (there's a 'Lower scan priority to reduce CPU usage' option that helps), but those who want to selectively disable real-time Internet security tasks while off-line, to squeeze the maximum performance from audio applications, can enable and disable the entire suite of OnGuard tasks using a single key-click. Make sure, also, that you remove the default Scheduled Task so that scans only occur when you ask for them.

As long as you always leave the OnGuard tasks running in the background while you're on-line, you shouldn't need to perform a more comprehensive scan of your hard drives very often. A full system scan using SDAV can take a long time, because it's relatively thorough, but for more general use the Intelli-Scan option will detect and disable active threats in just a few minutes.

You might ask why I was interested in investigating SDAV when I was already running a perfectly good freeware AV checker alongside the original Spyware Doctor. Well, the overall subscription for installation of SDAV on up to three computers, including new versions, unlimited smart updates and customer support for one year, is just £5 ($10) more than SD alone ($39.95 instead of $29.95), making the anti-virus components very good value for money. However, compared with running separate spyware and anti-virus utilities, the most practical day-to-day benefits for me were a speedier boot-up time for my PC and the time-saving aspect of only having to download updates for a single security tool instead of two.

Diagnose Drop-outs With DPC Latency Checker

Some musicians buy or build a new PC and achieve glitch-free, low-latency audio recording and playback fairly easily, while others experience occasional clicks and pops whose cause can be difficult to track down. I've devoted thousands of words to the many and varied causes of such interruptions (most notably in the SOS October 2006 PC Musician feature), but here's a handy tool that you can use to both check your system and investigate the causes of audio drop-outs.

 
If you find you can't run your audio interface at low latency, this DPC Latency tool may help you track down the cause of the problem. Thesycon's DPC Latency Checker (www.thesycon.de/eng/latency_check.shtml) is a Windows utility that polls the DPC (Deferred Procedure Call) latency once every second and displays the results in a horizontally scrolling graph. Hardware drivers issue periodic interrupts, and Windows deals with these as soon as it can, on a first come, first served basis (the DPC).

Unfortunately, some hardware spends too long dealing with its DPC routine. Readings under 500 microseconds (the green zone) are fine; those between 500 and 1000 microseconds (or 1ms) are borderline; and those beyond (the red zone) may result in audio interruptions. The beauty of this tool is it lets you spot occasional spikes above the norm and see how often they occur, which can be a great help in tracking down the culprit.

My PC typically measured around 50 microseconds, with occasional peak values of about 100 microseconds, but other musicians have reported occasional peaks of over 2000 microseconds (network adaptor cards are often the worst culprits, and particularly wireless ones). If you measure occasional high peaks you won't be able to run your audio interface at such a low latency, and should disable individual devices one at a time in Device Manager to find out which one is causing the problem.

Published May 200

Korg Kaossilator 2s: Make Music Anywhere

Audio Clicks and Pops

Clicks or pops in your audio? The cause may be hardware devices taking more than their share of interrupt time. Now there's a utility that checks for this, and a PC Notes survey amassing results that could help you track down the source of your problems.

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These DPC Latency results from an SOS forum user illustrate how enabling Wi-fi on a laptop PC can cripple audio performance.

Tracking down the possible causes of audio clicks and pops can be a thankless and frustrating task, so I was pleased last month to be able to recommend an easy-to-use utility for highlighting the presence of any rogue hardware device that occasionally takes more than its fair share of interrupt time (a classic cause of audio interruptions).

Thesycon's DPC Latency Checker (www.thesycon.de/eng/latency_check.shtml) has subsequently proved its worth many times over, especially once I'd started a 'DPC Latency Survey' on the SOS forums (/forum/showflat
.php?Cat=&Number=588140) and encouraged readers to submit their findings. In just three weeks the survey attracted over 100 posts and 4000 views, and there's already a lot to take in. Let's summarise the results so far.

DPC Latency Survey Results

DPC Latency Checker takes one measurement each second, providing a readout of current latency (which, on most PCs, is a relatively constant figure) plus the absolute maximum value recorded since you first launched the utility, which may occasionally spike to a considerably higher value on some systems, because of one specific hardware device.

The reason for such spikes needs to be tracked down to avoid possible audio interruptions (especially if you're using small audio buffer sizes). I've found the most effective approach after discovering spikes is using Device Manager in real time: select a possible rogue device in its list, right-click on it and select the Disable option. If the spikes are unaltered, you then re-enable the device and move onto the next one, until you find the culprit. When you find it, possible cures include updating its drivers, disabling the device each time you want to make some music, or creating a dual-boot system with it permanently disabled.

To get the ball rolling, I posted some baseline figures from my own well-behaved Intel Conroe E6600 2.4GHz dual-core PC running Windows XP with Service Pack 2, which typically measured 50 microseconds, with an occasional peak of up to 100 microseconds on my Internet-enabled partition. There was a smaller typical value of 24 microseconds, with an occasional peak up to 65 microseconds, for my music-only partition.

A hardware device needs to hang on to the interrupt for longer than 500 microseconds to enter the borderline area for audio interruptions, but these results do demonstrate that a stripped down, music-only PC is more likely to manage small buffer sizes (and thus lower audio latency) without audio glitching.

BIOS Beware!

Following last month's revelation in PC Notes that Intel BIOS updates could significantly improve audio performance at low latency with quad-core and octo-core PCs, the SOS forum's 'HL' reported that his Gigabyte P35DS4 v1.1 motherboard had lots of audio timing problems and stuck notes running Cubase with BIOS versions F7 to F11, and DPC Latency Checker reported regular peaks of at least 695 microseconds. Downgrading to BIOS version F4 or F5 resulted in typical values of just 30 to 40 microseconds.

Over the years, my advice on BIOS updates has always been to leave well alone unless the update specifically solves a problem you've been having, because if you ever had a power cut before the update was complete your PC might not be able to boot up at all afterwards. This new evidence provides a further reason to be cautious, so I would now, in addition, recommend that you run DPC Latency Checker before any BIOS update and again immediately afterwards. Don't be afraid to downgrade a BIOS if using the newer one results in latency spikes that weren't there before

Network & Wireless Spikes

Quite a few musicians have already discovered that disabling network adaptors (aka network cards, LAN adaptors or NICs), and particularly wireless (WLAN) versions, may cure audio clicks and pops, but DPC Latency Checker makes it far easier to spot possible symptoms early on. For instance, in this survey 'Phat Riffioso' discovered a regular peak of 405 microseconds every 15 seconds on his Asus P5WDH motherboard, which he traced to polling of an unused Ethernet port to check if any device had been plugged into it. The cure was either to disable the Ethernet port in Device Manager, or connect a device to it. 'Timo' also reduced his results from 100-440 typical and 550 maximum to 14-25 typical and 100 maximum just by disabling his onboard Realtek RTL8169/8110 Gigabit Ethernet, using Device Manager.

Another interesting finding from Timo was that moving his wireless mouse doubled his lower results to 30-130 microseconds typical and 226 maximum. Back in PC Notes August 2006 I reported on the Cursor XP mouse-pointer enhancement utility from Stardock (www.stardock.com), which managed to stop one musician's audio every time he moved his mouse, and started it again immediately he stopped moving it. So if you're replacing a hard-wired keyboard or mouse with a wireless model, or installing custom mouse or keyboard drivers, I now recommend using DPC Latency Checker before and after to check for spikes, as well as making an image file of your Windows partition so that you can easily backtrack if necessary.

US Court Allows Vista Lawsuit

A class-action lawsuit against Microsoft over the sale of PCs bearing 'Vista-capable' stickers has finally been approved to go forward by a US District Court judge, after customers were mistakenly encouraged to believe that PCs they bought during the Christmas 2006 holiday period could be easily upgraded to Vista when it shipped a month later. In fact, these PCs would only perform well with Vista Home Basic, which lacks many features, including the Aero Glass interface, Flip3D window switcher and Media Center.

However, of more general interest is a 158-page bundle of associated internal Microsoft emails that the same judge has recently been ordered unsealed as part of this case. These include reports that almost 30 percent of all early Microsoft Vista crashes logged by Microsoft were caused by Nvidia drivers. Microsoft themselves were in second place with nearly 18 percent. Most of these driver issues have since been solved, but this information once again underlines how driver performance can bring an operating system to its knees.
Other sobering figures for Vista can be found among the results of a recently published report from Forrester Research (www.forrester.com) covering 50,000 users across more than 2300 large enterprises, which indicate that during 2007 Windows XP was still being used by around 90 percent of businesses, with just 6.3 percent of users moving over to Vista, while Mac users have risen from 1.2 to 4.2 percent during the same period. With Windows 7 rumoured to have a release date in the second half of 2009, many businesses may already be considering bypassing Vista altogether.

Nobbled Notebooks

'Lachris' reported high average values of 200 microseconds on his Dell Precision M6300 notebook, with peaks of up to 2000 every 15 seconds, resulting in regular audio drop-outs, even at low CPU loads and with high buffer-size settings. These figures were subsequently confirmed by Dell on this model and their Latitude D630, and so far the only cure seems to be to buy a different laptop!

'Nuno' managed to get some good figures for his Dell notebook, but only after installing and configuring the third-party i8kfanGUI utility (www.diefer.de/i8kfan/index.html), which lets owners of certain Dell Inspiron, Latitude, Precision and Smartstep notebook models take manual control of their cooling fans to prevent real-time changes that cause audio drop-outs. Changes in fan rotation speed or CPU clock speed can prove an audio nightmare with some computer models (mostly notebooks). He also reported bad audio glitching on his particular notebook after installing M-Audio's Midisport 8x8 MIDI interface drivers, which was only resolved after a reformat and clean install of Windows.

 
Another forum user posted these figures showing the huge difference in DPC Latency between his Dell E520 desktop PC running Windows Vista and running XP.Once again, the moral is that before you install any new hardware drivers you should make a disk image file, run DPC Latency Checker for a few seconds to gauge its typical and peak results, then install your new hardware and check that the results haven't got significantly worse. If they have, and you're sure you have the latest drivers, at least you can restore the image.

Hasta La Vista

Continuing with Dell-related submissions, 'danf' posted the results from his Dell XPS M1330 notebook running Windows Vista (typical 100/200, but occasional spikes to 1000+). He managed to reduce these figures to between 60 and 70, with occasional spikes to 220, simply by installing Windows XP instead. Danny Bullo got an even bigger improvement with his HP Pavilion dv9500t laptop, which gave spikes of up to 3000 microseconds with Vista, but a very low 50-microsecond peak with XP! 'Muied Lumens' then switched to XP after a year of tweaking Vista, reducing almost continuous 1000-microsecond readings to a mere 12-15, with peaks of just 100 microseconds.

While we can't actually conclude that Windows Vista itself is to blame for all these high readings (hardware drivers and, in particular, graphics-device drivers are currently under suspicion) the fact remains that by switching to Windows XP these three musicians solved their audio problems. Perhaps Vista performance will simply leap ahead when more optimised hardware drivers have been developed for certain rogue devices.
DPC Latency Checker may also help us identify problems caused by rogue Firewire controller chips and badly written audio interface drivers, so please report anything unusual you find!

Published June 2008

The 2015 NAMM Show: Rich Formidoni talks about the ARP ODYSSEY, MS-20m and SQ-1

Q. Do I really need 24-bit recording?

Is 24-bit recording any better than 16-bit recording in a home studio, given that the only qualitative difference is (apparently) the noise floor? In 99.9 percent of home studios, this difference will be masked by the ambient noise of the studio, and by preamp or mic self-noise. From what I've learnt from Hugh's articles on digital audio, I'm sceptical. Is it really worth me switching from 16-bit to 24-bit recording?

 
SOS Forum Post
 
 
Recording at 24-bit increases the available headroom in the system, although this may not offer any practical benefits for signals recorded with microphones in an average home studio.  

Technical Editor Hugh Robjohns replies: The characteristic defined by the word length is the available dynamic range, or where the noise floor sits in relation to the peak level. A properly dithered 16-bit system provides a dynamic range of around 93dB, while a mid-budget 24-bit system should be able to deliver something around
115dB or so. The implication, then, is that you could work with about 20dB more headroom in a 24-bit system without any increase in the apparent system noise over the 16-bit mode. For most people, the additional computer overhead of operating with 24-bit word lengths instead of 16 has a negligible effect on processing power, and the additional file-storage requirements are inconsequential, given the size of modern hard disks, so the lower system noise and greater headroom margins are a welcome benefit for a negligible cost.

Having said all that, you are right in that for most people recording at home, the recording noise floor is almost always defined by the ambient room noise rather than the digital system noise. Therefore, you can usually work with reasonable headroom margins at 16-bit word lengths while maintaining the ambient noise floor comfortably above the system noise floor. In other words, the dynamic range of the source recordings is typically far less than the capability of a 16-bit recording system. Where that changes is if you record electrical instruments such as guitars and keyboards via DIs. In these cases, the noise floor is often far lower than that possible when recording with microphones in a home studio, so the dynamic range of a 16-bit system may start to become a limitation.

The bottom line is, if you are happy with the results you get with 16-bit, there may be no benefit in changing to 24. However, most people do find that 24-bit working enables greater headroom margins without noise penalties, and that in turn makes recording less stressful and mixing rather easier. Personally, I can see no negative side to recording at 24-bit, and only positive benefits.

Published June 2008

The 2015 NAMM Show: OrKeystra joins us in sharing their love for the new Korg Kronos

Q. How can I use my sampler with Logic?

I've recently upgraded to an Intel iMac with Logic Express 7, and have been experiencing a few problems while transferring onto this format. My previous setup was an old Mac Performa running Opcode's Studio Vision. I was using this with an Emu ESI2000 sampler. I had been running all of my audio in the sampler using samples, recorded audio and various instrument and synth presets. Working this way I developed a style and sound that I really like, especially with drums, but I had real problems mixing and finishing tracks.

 

The ESI2000 (top) can be fitted with a Turbo board to add extra outputs, if you want to record multiple instruments at once, but a soft sampler, such as Logic's own EXS24 or Native Instruments' Kontakt, can be much easier to use and more powerful than older hardware samplers. You can easily trigger imported samples from such soft samplers.I'm still using the sampler to make beats and then I'm trying to transfer them into Logic. I'm finding this a really awkward process, as my audio always plays back in a different place. So, if I record different tracks individually, it's always difficult to get the timing spot on. How can I change this? Also, is there a way that I can record my drum samples into Logic, and then play them in Logic as MIDI, such as I would when using a soft synth or internal instrument? I don't seem to have a software sampler with Logic Express, so maybe I need the new Logic 8? The problem at the moment is that I want flexibility to edit the MIDI data right up to the final stages — editing audio is obviously much more restrictive with things like beat programming. I also want to be able to use the production facilities on offer in Logic while using MIDI programming.

Adam Hurst

Reviews Editor Matt Houghton replies:
If you're finding that the timing is not sufficiently consistent to record the different parts in separate takes, you could try recording the different sampled instruments simultaneously to different tracks in Logic. If you do that before you add other elements to your project, they'll all be in time and you can calculate the tempo and adjust the project in Logic to suit before adding other elements. This will require your sampler to have sufficient outputs (there are four, but it is expandable to eight, with further digital channels available, if you have the Turbo board installed), and you'll need a soundcard with the same number of inputs.

I'd recommend getting a soft sampler, though, as you'll eventually find it much easier to use, and much more powerful than your old system — and if you want to you can import all your old samples from the Emu into the soft sampler and trigger them in there instead.

Logic Express 8 includes the EXS24 soft sampler, whereas v7 only includes the cut-down EXS24P (which will only play back pre-programmed libraries). EXS24 is also part of the 'full fat' version of Logic, so you'll need to upgrade to Logic Express 8, Logic Pro 7 or Logic 8 if you want to use EXS24. Alternatively, you could buy a third-party soft sampler like Native Instruments' Kontakt, or maybe even download a freeware sampler: try the database at www.kvraudio.com if you want to search for one.

In most soft samplers, you need to record your samples in as WAVs or AIFFs using a program like Logic (any audio recording/editing software will do) and then load the samples into the sampler. We've run a number of workshops on EXS24 in the past, and the basics haven't changed a great deal, so you might find it helpful to read a few of them if you're new to all this. Try this one to get started: www.soundonsound.com/sos/aug02/articles/exs24.asp.

The 2015 NAMM Show: Rie Tsuji talks about the new Korg Kronos

Q. Where should I put my violas?

I'm going to be recording a string section. I'll be recording the section using both close-miking and a Decca tree in order to have maximum recording flexibility. We'll be doubling a lot of sections and, further down the line, adding a 10-piece brass section and some samples to make the recording sound fuller. Can you please advise me as to how I can place my 14 string players in order to get the most out of them? I have five first violins, three second violins, two violas, two cellos and two double basses. As I only have two violas, I'm worried that placing my players in their normal orchestral positions will leave a big gap in the middle of the formation, where the violas would be placed. I'm trying to emulate a large orchestra and have been mulling over whether a wider spread of players will help or not. The danger is that the players will not blend, and instead sound like individual players, which must be avoided. What's your advice here?

Via email
 
SOS contributor Mike Senior replies:
 
 
With only two violas in the section, there could be problems with regards to the mix balance. If you have time, you could try overdubbing extra violas to make up for the shortfall. 

If this orchestral arrangement is designed to complement some kind of pop or rock track, then leaving a bit of a hole in the middle and over-emphasising the width of the section might be just what you need. This will keep the strings out of the way of other more important parts of the track, such as lead vocals. Plus, with everything else going on, you probably won't notice a few blend problems in the grand scheme of things.

However, it sounds to me more like you're trying to mock up an orchestral score for a film or something similar, in which case the situation's a bit different. You might find that you have mix-balance problems with only a couple of violas. A lot depends on the specific players, instruments, and arrangement, but the numbers are a little against you. If it's just the stereo image you're worried about (and you're not happy for this hole to be filled by the winds and brass), then move the violas a little closer to the cellos, and bring the whole string formation tighter together. If balance is still the problem, then you're right that pushing the close mic nearer to the violas, to compensate for their minority, could easily affect the blend. If you have time, you could try overdubbing the violas as a separate part to give you a fall-back option. (If the lack of other parts would make it impossible for the violas to coordinate their part solo, then put in a token player on whichever other sections are required to make it possible.)

The important thing to remember when doing this is to record the Decca tree alongside the viola close mics: this will give you more viola ambience to help gain the illusion of more viola players. That said, if you're going to be including samples anyway, you might find that you can deal with any imbalance at that stage. A pair of violas, especially if doubled, should be enough to disguise quite a lot of extra sampled viola parts. If there is an exposed viola line that you're worried about somewhere, but you still want to prop things up with more samples, then adjust the arrangement a little in that spot to double the violas with some of the violins or cellos. If all else fails, and you have to turn up the viola close mics in a way that affects the blend, then you might be able to compensate with a little added reverb on those mics.

However, If you're aiming to capture a very natural sound, more akin to a classical-style ensemble, then you have to ask yourself how much real stereo positioning you'd actually hear from a couple of dozen rows back in a concert hall. I'd question the idea of widening things too much if that's your target sound.

Published June 2008

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Q. Do I need a professional setup to record on my laptop?

I want to mic up my bodhran so that I can record what I am doing in Audacity, then play it back in order to hear where I'm going wrong, with the hope of improving my technique. One of the mics I've been recommended is an AKG C418 clip-on model, but I'm told it needs phantom power. What's this phantom power thing, and do you think I could achieve it without a fully professional recording setup?

I'm intending to record and play back from my PC laptop, but I can record to cassette tape if necessary. I also have a hi-fi amp, which I currently use to transfer the odd bit from vinyl and cassette into Audacity on my computer, should it be of use.

John Blackwell


 

You don't need expensive gear to mic up the majority of instruments. This setup of an AKG C518 clip-on mic and a Yamaha MG102C mixer costs under £300, and is capable of producing great results. You can spend even less money if you go for budget brands...  

News Editor Chris Mayes-Wright replies: Phantom power is used to power specific types of microphones. Models such as the omnipresent Shure SM58 and other common handheld vocal mics don't need phantom power because they are dynamic — they operate like a speaker working in reverse, turning acoustic vibrations into electronic signals.

The AKG C418 (superseded by the new C518, pictured below) is a very capable condenser microphone that's often used for close-miking drum or brass instruments, due to its handy clip-on nature and gentle low-frequency roll-off. Its capsule (the bit that moves when sound waves hit it) is electrically charged, and requires a power supply to deliver this charge. So phantom power is sent from the mixer (or whatever the mic is plugged in to), down the microphone cable, and is used to polarise the capsule and also power other internal circuitry.

Phantom power is found on almost all modern mixing desks, professional or not. It's normally indicated by a button with '+48V' or 'phantom' next to it. The nominal supply voltage for phantom power is 48V, but some phantom power circuits only produce figures around the 30V mark.

To connect a mic to a phantom power supply, simply plug it in to the XLR inputs on the mixing desk, and switch the phantom power on. It's not good practice to leave phantom power on then plug the mic in, as this can cause surges when the cable makes an electrical contact with the microphone.

Hopefully that gives you a basic idea about what phantom power is. All you need now is some advice on how to acquire it! A cheap mixing desk is the obvious thing to go for, and you can get a usable one for under £50 and a more-than-half-decent one for around £70 — the Yamaha MG102C (pictured) was on sale for £69 at the time of going to press. At both of these price points, you'll get something that can be used for applications other than powering your AKG C418 (or C518, if you decide to purchase the new model), such as mixing three or four instruments and feeding their signal into a PA system. The cheapest mixers don't have phantom power, so make sure you get one that does! Any such mixer will have outputs that you can send to your cassette recorder, PC laptop, hi-fi amp, or main-stage PA system, for when you've perfected your playing technique!

If you just want to record into Audacity, and you don't need the other mixer paraphernalia, you can get an external audio interface — basically a soundcard that connects to your computer via USB or Firewire. If going down this route, make sure you choose an interface that has at least one microphone preamplifier and phantom power, for obvious reasons. All you have to do to record on to your computer is install the software drivers for the interface and make sure everything's connected correctly. Then, Audacity should be able to 'see' your incoming signal from the connected equipment. Interfaces cost anywhere from £40 to £1500, but you can get a decent two-channel model for just over £100.

As another alternative, you could get a single-channel microphone preampwith a line-level output that you could plug straight into your hi-fi amp or PC line input. Prices for these start at around £60.

Personally, I'd go for the first option I mentioned: a small mixer. It's the most flexible and cost-effective approach.


Published January 2008

Korg Pa600QT Professional Arranger Performance

Q. Why so many digital audio formats, and what are they for?

There are so many ways of connecting equipment these days, such as S/PDIF, ADAT, AES-EBU and MADI, not forgetting good old analogue. What are all the digital connections for?

 
SOS Forum post

Technical Editor Hugh Robjohns replies: Back in the '80s and '90s, there were dozens of manufacturer-specific digital interfaces, such as Yamaha Y1 and Y2, Melco, TDIF, ADAT, SDIF2, R-Bus and many more, and none of them could be connected together. It was a complete nightmare!

In order to make 'going digital' a practical option, the Audio Engineering Society (AES) and the European Broadcast Union (EBU) put their collective heads together and came up with two generic, open-source digital interfaces: one for stereo and one for multi-channel audio, the latter of which was originally intended to link multitrack recorders to large consoles. These were called AES-EBU (now more commonly referred to by the AES standards document number, AES3), and MADI (Multi-channel Audio Digital Interface).

 
A comparison chart showing different types of digital audio protocols. 

AES3 was a bodge in the engineering sense, but the use of apparently standard mic cables and connectors made it a familiar-looking interface that reduced the fear and cost of 'going digital'.

The original MADI specification essentially carried 56 channels, made up of 28 AES3 stereo pairs transmitted serially. A later revision called MADI-X catered for 64 channels and is in widespread use today in applications such as connecting stage boxes to digital desks, linking Outside Broadcast trucks, and connecting the infrastructure in digital studio complexes.

Today, AES3 is the preferred interface format for professional stereo applications, although there is a noticeable trend towards the AES3-id format which uses unbalanced BNC connectors and 75Ω video cables rather than balanced XLR connectors and 110Ω cables. AES3-id is a much better-engineered interface, and is far more space-efficient. AES3 digits run with a fundamental frequency of 1.5MHz, with strong harmonics all the way up to 10MHz and more. Video cable and connectors are far better suited to handling those kinds of frequencies than manky old mic cables, and AES3-id works more reliably over greater distances, with less jitter as a result.

Having designed a very versatile and effective digital interface, and all the hardware chips to drive and receive it, Sony and Philips took the opportunity to use the same thing for domestic applications and called it S/PDIF, with coaxial (phono) and Toslink (optical) interfaces. The nitty gritty of the auxiliary information and metadata carried by AES3 and S/PDIF are slightly different, but the basic structure and audio formatting are identical, and you can normally interconnect AES3 and S/PDIF with little problem. S/PDIF is electrically almost identical to AES3-id.

In terms of the actual interface properties, AES3 runs balanced signals with a nominal 7V peak-to-peak swing, feeding a receiver with a minimum sensitivity of 200mV. Because the signal starts so big, it tends to go a long way (more than 100 metres) even on nasty mic cables. Put it into decent low-capacitance 110Ω cable and it will easily travel 300 metres. AES3-id is unbalanced and starts at 1V. The receiver sensitivity is the same 200mV, while S/PDIF is also unbalanced and starts at about 0.5V. The receiver sensitivity is also 200mV. The lower starting voltage is why S/PDIF doesn't travel very far.

There are several eight-channel AES3 interfaces, most using 25-pin D-Sub connectors. Sadly, there are lots of different incompatible pin-outs: Yamaha, Tascam, Genex and Euphonix, to name a few. But the Yamaha and Tascam formats are the most prevalent.

Yet another variant of AES3 is called AES42. This still uses XLRs and balanced cable, and the data is encoded in exactly the same way as AES3, but it is intended for carrying the output of digital microphones. The critical difference is that an AES42 input socket provides 10 Volts of phantom power, and that power is modulated in a specific way to allow remote control and digital clocking of the microphone. It is an agreed format that has been adopted by Neumann, Sennheiser, Schoeps and others, and will start becoming a common feature on digital consoles and professional recording interfaces.

In terms of other digital interfaces, Tascam's TDIF is virtually dead, but ADAT is alive and well and in widespread use. ADAT uses the same Toslink fibres and connectors as S/PDIF, but with a different data-stream structure to carry eight channels.

In addition to MADI for high numbers of channels, we also now have the new AES50 SuperMAC and HyperMAC audio networking interfaces (originally developed by Sony Oxford in the UK and now owned by Telex Communications under the Klark Teknik brand).

SuperMAC provides 48 channels bi-directionally over Cat 5 cable, while HyperMAC provides up to 384 channels bi-directionally over Cat 5 or Cat 6 or fibre. The signal format includes embedded clocks in the same way that AES3 does.

Published February 2008

The 2015 NAMM Show: Dirty Loops talks about the new Korg Kronos

Q. How do I record brass in the home studio?

The crack team of Paul White and Hugh Robjohns have travelled the world solving readers' problems. Here, they down the Hob Nobs and answer some of your recording queries in our Q&A mini-series, Sound Advice.

Paul: The most common brass instruments you'll find recorded in home studios are trumpets, trombones and saxophones (the sax is technically a woodwind instrument, as it has a reed, but is so commonly found in 'brass' or 'horn' sections). Recording brass is something I do relatively infrequently but it has never really presented any serious problems. As long as you make some effort to get the mic where it sounds best, and make the player aware that if they move around, the timbre of the sound will change dramatically due to the extreme directionality of brass instruments, the rest is pretty easy!





 
Hugh: In the case of trumpets and trombones, the majority of sound comes from and projects in the direction of the instrument's bell. The 'polar pattern' of the instrument becomes increasingly directional with rising frequency. If you stand behind a trumpet you'll hear very little direct sound and no HF components at all. If you move to the side you'll pick up the lowest frequency components, but it is only directly in front that you'll hear the higher frequency components and harmonics (above about 4kHz).

Paul: So for the sharpest, crispest sound you'll need to place a mic directly in front of the bell ­ either on a fixed stand or clipped directly to the instrument itself. If you're concerned about room reflections, then some suspended duvets behind and to either side of the player will help damp things down. As is often the case, if the room doesn't flatter the instrument, then it is better to damp out as much of the room as you can and then replace it with a more sympathetic reverb when you come to mix.

Hugh: You need to bear in mind that brass instruments are designed to be loud. Trumpets have been measured at four metres to produce over 96dB SPL, and around 130dB SPL just 0.5m from the bell. A trombone is roughly 5dB louder still. So microphone choice means finding a microphone that can accommodate huge peak levels without excessive distortion.

Paul: But every cloud has a silver lining, so they say, and the high SPLs of brass instruments mean that even noisy computers in the same room are so far below the levels of a typical brass instrument, to the extent that, unless you have the mic set up right next to the computer, noise isn't going to be a problem. Again, some acoustic treatment may be necessary, though, because while you can drown out background noise, you can't drown out sound reflections — they get louder as the instrument does!

Hugh: When it comes to mic choice, a typical studio approach for pop music would be to use classic large-diaphragm condensers such as the Neumann U87 or AKG C414, operating with pads switched in and placed within half a metre of the bell. In practice, most large-diaphragm mics will work fine provided they can cope with the peak SPLs. Failing that, dynamic mics are always a reliable choice that will cope with the SPLs without trouble. Ribbons are also popular, both for pop and classical brass recordings. The Coles 4038 and AEA R44 are favourites, but you need to place them no closer than a metre and use a pop screen to prevent wind blasts popping the diaphragm.

Paul: Unless you have headphones with very good isolation, you may not be able to hear the effect of mic movement when working 'live', so recording a test section while moving the mic and describing the mic positions into the mic as you do it may work better. Further to what Hugh suggested about choosing a mic that can handle high SPLs, I suggest that you leave adequate headroom at your mic preamp and DAW input, as the quirky waveforms produced by some wind instruments can produce misleading meter readings on some less-sophisticated metering systems.

Hugh: The saxophone needs to be treated a little differently, because it generates sound in the same way as that conventional woodwind instruments do: along the full length of its body. This means that a different miking technique is appropriate: the mic should be placed to 'hear' the whole body of the instrument. However, in pop music, it has long been fashionable to mic tenor and baritone saxes very close to their upturned bells (as in the picture), and this captures a very specific kind of sound. It's not the natural sound of the instrument as heard in the room at a distance, but a bright, raspy sound that everyone recognises and now expects.

Paul: Some final advice: when recording, pay attention to the sound quality. It changes as the instrument (and player) warm up, and as the instrument inevitably fills with spit! It pays to give the player plenty of opportunity to clear the instrument and rest the lips, particularly if the playing involves a lot of stabs and high notes. Brass players can tire quickly — these aren't easy instruments to play — so be realistic in how much recording and overdubbing you can do.

Published March 2008

The Korg Launch Event 2015

Q. How do I record small percussion instruments?

By Hugh Robjohns & Paul White

The crack team of Paul White and Hugh Robjohns have travelled the world solving readers' problems. Here, they down the Hob Nobs and answer some of your recording queries in our Q&A mini-series, Sound Advice.

Hugh: Recording hand percussion is often more challenging than it would initially appear to be. Part of the problem is that often the sound pressure levels aren't that high — so close-miking appears to be sensible approach to take — yet a lot of physical movement is involved in playing the instrument in question, making distant mic placement seem like a better option.

 
Paul: Whatever you're recording — from balaphons to finger cymbals and thumb pianos — you will need a microphone that is able to deal with a very wide range of frequencies. As Hugh mentioned, some instruments are very quiet, so a capacitor microphone with a low self-noise figure (typically less than about 17dB EIN) would be a practical option in most cases.

Hugh: Agreed, percussion obviously involves a lot of fast transients, and the detail of the sound is conveyed by those transients, so a responsive microphone such as a capacitor is a must. But ribbon mics are enjoying renewed popularity, thanks to the new cheap components flooding the market. These tend to sound smoother and more natural than capacitor mics, without any resonant emphasis at the high end (which can be an issue with tambourines, for example). However, most have a figure-of-eight polar pattern, which will result in more room pickup than, for example, a cardioid mic. Some ribbons are designed with 'bright' and 'dark'-sounding sides, so some experimentation may be appropriate to see what complements the percussive sound best.

Paul: Deciding whether the sound of the room enhances or degrades the recorded signal is something that you will have to do after listening to what's coming from the mics. Generally speaking, domestic rooms tend to sound boxy and add little to the life of the sound. Therefore, you may be better off keeping the recording fairly dry and then adding ambient reverb (predominantly early reflections) when you mix. If you find that you're getting too much of the room sound in your recorded signal, you can place a broadband absorber, such as a commercially available filter or some thick duvets, behind the mic. If you're using Hugh's suggestion of a bi-polar ribbon mic, this will also help to negate the contribution from its rear lobe.

Hugh: Mic positioning varies from instrument to instrument, but my general rule for capturing a natural sound is not to bring the mic closer than the longest dimension of that part of the instruments that produces sound. In the case of a drum, this would be the head diameter, though as with all drums, you can mic them very close up if that produces a more useful sound, even though it may not be as accurate as miking from a greater distance. It's always best to search for the sweet spot, but as a fallback position, you can usually capture a decent sound by miking over the player's shoulder, providing the instrument sounds good to the player.

Paul: Once you're happy with the sound you're getting from your mic, there are various things to consider when actually recording it. Because of the inherent nature of most hand percussion, often involving loud and brief transients, ensure you record with a generous headroom margin — I would suggest at least 12dB. In many cases, there will be little low-frequency content, and filtering off the low end during recording can help reduce unwanted room colorations quite effectively.

Hugh: Absolute rhythmic accuracy is usually of prime importance with hand percussion, and if the performer's abilities are limited (playing hand percussion accurately for a three-minute track is extremely difficult and tiring), then there isn't much shame in identifying a bar, or a couple of bars, that work well, and then copying and pasting them as necessary in to the track.

Paul: In the event that you record a percussion part that just doesn't stand out as we'll as you'd hoped, I recommend using an enhancer, my favourite being the Noveltech Character plug-in for TC Electronic's Powercore platform.


Published January 2008

Korg’s new Kronos at Dubway Studios in NYC

Q. How do you know what EQ frequencies to cut?

Thanks for featuring my band Imprint in the Mix Rescue article of SOS January 2008; we were really happy with the results that Mike Senior achieved. But whenever I read such articles, there are always mentions of very specific EQ cuts or boosts at very specific frequencies. In Mike's write-up on our mix, he says "drum harmonics were dominating at the low end — cuts of 3-4dB at 47Hz, 59Hz, 96Hz and 174Hz were all that were needed". How are these frequencies arrived at? Is it just a case of knowing the problem frequencies, is it with some sort of spectrum analyser, or it is trial and error? Also, how are the cuts done — is it just using a multi-band EQ?

I think this is the weakest part of my mixing side, and is probably the part I want to work on next, but I am not really sure where to start.

Blink (Imprint)

SOS contributor Mike Senior replies:
Drum samples, such as the ones in use in your track, don't usually change pitch, so you can use very narrow EQ cuts to home in on individual harmonics of the sound, and this is great for dealing with unwanted resonances in particular: a ringing sound at a particular frequency, for example. The problem with over-prominent individual harmonics is that they stop you fading up all the other harmonics of the sound far enough in the mix. By the time the sound as a whole is audible enough, those little resonant frequencies poke out too far. Very narrow EQ cuts make very little difference to the rest of the sound (the narrower the better really), so they're one of my favourite tactics when I can get away with them. They won't work on melodic instruments, though, because the harmonics move around as the pitch changes.


In terms of which EQ to use, it doesn't matter at all, as long as you have a Q/resonance/bandwidth control and you can accurately enter frequencies — particularly at the low end, where 1Hz can mean a noticeable difference in pitch.

From memory, in your particular case I seem to remember that the kick part had two pitches to it, and I felt that both had low harmonics to them which were a little too prominent. This made them boom and hang on too long for my liking, especially as I knew that the effects and everything else would need space to move in. One of the pitches was more problematic than the other, hence the rough harmonic relations between some of the cuts. Still, the cuts weren't particularly severe. I normally find myself cutting more severely when using this technique. Here these notches were effectively just 'pulling down the fader a bit' for those individual kick harmonics without affecting the level of the kick as a whole.

The way I found those particular frequencies is by using a fairly well-known trick. Take a peaking filter and ratchet up the resonance (or Q value) to maximum. If you then apply a big gain boost with that filter, it acts as a kind of audio magnifying glass, picking out very narrow frequency bands and even individual harmonics of a pitched sound. If you sweep it around the frequency range for a while you'll soon get an idea of where the problem frequencies lie, at which point you can reverse the gain control setting to cut instead of boost.
You do need to be careful with this approach, though, because it's easy to be tricked into hearing problems that aren't actually there. What I do as a reality check is, once I've finished setting up the cut in question, I bypass it for a few seconds, and listen for the thing that I don't like. Only when I have it pinned down in my head do I re-engage the filter. That way I know for sure whether my EQ setting is doing the job I initially wanted it to. Funnily enough, I have a vague recollection that I tricked myself a couple of times with that very kick sound (maybe by initially setting the harmonic relations by eye rather than by ear), and I did come back to it and tweak it a couple of times as things progressed. The exact cuts at the two lowest frequencies in particular had to be quite finely judged — if you follow the fader analogy I made above, it makes sense that you'd need to adjust them as finely as you would any other fader.

My bottom line with EQ is that if I can't find a main fader level that I'm happy with for a given instrument, it's often because I actually need more than one fader for different frequency regions of that instrument. Using EQ makes a lot more sense if you think of it as just giving you these extra faders. In some senses, a graphic equaliser makes this easiest to visualise, but a parametric EQ will probably sound better, and it will offer more accurate control, especially for things such as killing pitched resonances. So I usually use a parametric.

Published March 2008

NAMM 2015: Toshi Hiketa performs with Nutube

PC Freeware Sequencers & Editors

By Martin Walker

Not only are these applications easier to use than commercial packages that can be confusingly feature heavy, they're also extremely easy on the wallet...

Musicians new to PC sequencing often feel overwhelmed by the sheer number of features provided by the flagship versions of modern MIDI + Audio sequencing packages, and are on the lookout for easier-to-use applications. Most commercial packages have 'entry-level' versions that are cheaper but may not appear much simpler to the novice, since they merely cap the maximum numbers of simultaneous audio/MIDI tracks, soft synths and insert plug-ins, while their menus remain awash with options. Even some freeware sequencers manage to baffle the newcomer unused to concepts like automation, external clocking, and so on.

On the other hand, not every new PC musician wants to create all their songs using construction-kit software plus thousands of bundled audio loops! There certainly seems to be a demand for serious creative applications that have fewer options and are easier to use, and this is what set me off on my quest to find out what you could achieve with simpler freeware sequencers and audio editors.

Setting The Scene

It never ceases to amaze me how many talented people find the time to develop and continue to refine the amazing freeware applications available, and are generous enough to release them for all the world to use without charge. I'm also surprised at just how many good freeware sequencers are available. Some are free 'lite' versions of more sophisticated products, while others are free for personal, educational and non-commercial use, but you're encouraged to pay a suggested (small) fee for a commercial licence (in other words, if you end up making money from music you create using it).

Other products are simply classified as donationware: you can download and use them free of charge, but users are encouraged to send a modest financial contribution to help pay the developer's bills and encourage further development (often using Paypal, the most popular way for anyone with an email account to securely send or receive on-line payments using their credit card or bank account).

I must mention one particular sequencer application in passing, given the shock waves it has managed to send through the audio community for its slick, professional interface and huge array of functions. Strictly speaking, Reaper (www.cockos.com) is shareware, but given that you can download the full, un-crippled version to try out, some might consider that other freeware sequencers would now be dismissed out of hand. However, this hasn't proved to be the case. As I said in our recent in-depth SOS review, I was most impressed with Reaper, but (like some other musicians) found some of its features initially confusing, and many novices seem to end up bewildered by the number of choices available. The applications I've chosen to feature in this round-up are those that are easy to get into, yet capable enough to accomplish a variety of serious musical tasks. Along the way I did have to discard a few that were either confusing or unreliable: some seem to remain in Beta versions for several years.

Quick & Fun Freebies

If you fancy a quick, fun approach to making sampled music, why not download Richard Spindler's Gungirl Sequencer (http://ggseq.sourceforge.net/HomePage)? Its approach is simplicity itself — you just use the left-hand folder tree to click on your desired sample folder, and drag files from this folder directly onto any of the eight tracks that appear by default beneath the timeline (although you can add and delete tracks as you need). You can loop any section of the timeline while working on your songs, and each track has its own volume control and mute button, while a global slider controls the overall output level.

You can set the 'snap' value in BPM, frames or seconds, so your samples line up easily on the beat, and to help you do this there's an optional info window, when you audition your samples, that displays their length. Once they're positioned on the screen-tracks, you can drag-copy and move your samples singly, or en masse by rubber-banding a box around them, add fades or control their volume envelopes, and even open up a simple sample editor where you can adjust start and end points and apply time-stretching to make multiple files at different tempos fit your songs. You can even export and import packages of songs, plus their associated samples, so you can collaborate with friends. Professionals might mock, but Gungirl is fun, and it's free.

If you want a very simple and compact pattern-based MIDI sequencer, PQN Audio's VstSeq (http://pquenin.free.fr/pqnaudio/vstseq) is a 132KB download that lets you enter and edit notes manually in its pattern windows, which can have between one and eight measures, each of between two and 32 steps, at a tempo of between 30 and 300bpm. You can send its MIDI output to any of four VST Instruments, and then either export your completed pattern as MIDI files to another sequencer, or render them as completed WAV files.

Kristal Audio Engine

The Kristal Audio Engine (www.kreatives.org/kristal) is an audio-only sequencer in a state of flux between freeware and shareware status. The freeware version 1.0.1 was developed between 2003 and 2004 and is still available for free personal and educational use, but commercial users are asked for a modest 24.90 Euros for a single-user licence. Meanwhile, its lead developer created Kristal Labs Software Ltd in 2006, in order to develop a new commercial product, code-named K2, that looks to be nearing completion (you can pre-register to be informed by email when it's ready).


Kristal Audio Engine: With an easy-to-use interface, plus support for both ASIO drivers and VST-format plug-ins, the capable Kristal Audio Engine provides easily enough features for musicians who only require audio recording and playback.

I found the freeware version refreshingly straightforward. It supports either MME/WDM or ASIO (for lower latency) drivers and offers 16 audio tracks at sample rates of between 44.1kHz and 192kHz, which is quite enough for most musicians who want to record a band or their own music using acoustic/electric instruments, and who don't need MIDI or soft synths. I suspect that KAE may also appeal to musicians who record with a hardware multitrack, yet want to mix on PC.

The main workspace for recording, editing and arranging is termed the Kristal Waver, and is a variation on the familiar 'arrange' page, with horizontal tracks, each containing one or more recorded or imported parts and each with its own 'Inspector'-like panel on the left, containing record, monitor and mono/stereo switches. The usual click/drag move and copy functions are available for parts, while you can alter the start and end points, level and fade in/out times of parts using their graphic 'handles'.

Across the top of this page is a tool bar containing select, cut, and glue tools, multi-stage undo/redo, auto-scroll and snap-to-grid options, then an info line providing details of the currently selected part, a Zoom strip giving a graphic overview of the entire song, and a Time Ruler calibrated in bars and beats, seconds or samples, where you define loops and so on.

The transport panel provides another familiar set of controls, including a set of 'tape transport' buttons, left/right locator displays, optional metronome, BPM and time-signature readouts, and (very handy for band recordings) a pre-count function using the metronome, to give you time to prepare yourself before a take.
A lot of the creative work goes on in the floating Mixer window. Each of the 16 tracks has its own channel strip with fader, pan, meter, mute/solo buttons, an integral three-band parametric EQ and two insert slots into which you can load any VST plug-in in your collection (and DX ones, if you first install a suitable DX-to-VST wrapper utility). The stereo Master channels provide three insert slots, and the package also includes a few of its own VST-format plug-ins: the three-band parametric EQ again; a chorus; a reverb with a smooth tail; the Kristaliser limiter/distortion; and the surprisingly versatile Multidelay.



SEQ24: If you want a MIDI-only sequencer optimised for real-time live performances using a clutch of hardware synths, this could be just the job.

The only aspect of KAE that I found initially confusing was the mixer's Audio Input slots. There are four available, each of which can host its own Kristal Waver arrange window or a so-called 'Live IN' plug-in. Strictly speaking, the latter are not plug-ins at all, but mini-mixer windows where you can combine up to eight mono or stereo input signals (assuming you have a corresponding number of inputs on your audio interface), adjust their relative levels and reduce them to a single mono/stereo output signal that you route 'live' (subject to normal latency delays) through the EQ and VST effects in the main KAE mixer.

If, instead, you select 'Kristal Waver' for an Input slot, another arrange window appears, so a theoretical 64 tracks are available across the four Audio Input slots (16 for each Waver window), and as soon as you record/import audio data into any tracks in any Waver window they become automatically connected to a mixer channel. However, although playback across these multiple Waver windows remains in perfect sync, there are no 'tile windows' functions to help you visually line them up, and since the mixer only supports a maximum of 16 channels anyway, I suggest you stick to using one Waver window and avoid the extra options and confusion.

Apart from this, I found using Kristal Audio Engine a very pleasurable experience, and there are some helpful tutorials in the HTML manual on Project Management, Recording, Mixdown and Export, plus how to use VST effects. Some potential users may eventually miss automation and it will be interesting to see what additional features appear in K2, but I suspect that most potential users will be quite happy with what there is.

Other Free Sequencers To Try

Although some sequencers at first appear to be freeware, some turn out to be demo versions of shareware products, either with the save functions disabled, restrictions on song time, or a sub-set of enabled features. Here are some you might like to try out that are either completely free or donationware:

• TuxGuitar: If you're a guitarist, why not check out the freeware TuxGuitar sequencer? It offers a set of features that have been specially tailored for guitar players, as well as a piano-roll editor and a lyric editor.TuxGuitar (www.tuxguitar.com.ar) is a multitrack tablature editor and player with special features for the guitarist, including support for various effects (bend, slide, vibrato, hammer-on/pull-off, grace notes, harmonics and so on), plus a score viewer, piano and lyric editors.
• Sequitur (www.angryredplanet.com) offers "dynamic MIDI-oriented music editing and real-time manipulation" as well as an elegant interface and an interesting 'Echosystem' tool for generating patterns. However, it's still in an early Beta stage and I experienced several crashes while using it, so take care if you try it out.
• The freeware version of Anvil Studio (www.anvilstudio.com) is a more traditional MIDI-based sequencer offering comprehensive staff, lyric, piano-roll, drum, loops, audio and event editors, along with support for a single mono/stereo audio track. This may well suit those with 'read the dots' ability.
• Maize Studio (www.maizesoft.cn) is a 'modular live audio environment' where you build audio devices, connect them in Design View (a virtual patchbay) and see their front panels in Device View. It supports ASIO drivers and VST plug-ins and includes a disk-streaming sampler and audio player.
• Digital Sound Planet (www.digitalsoundplanet.com) have a freeware version of their Quartz Audiomaster, but I suspect that many SOS readers will find its four audio tracks and maximum 16-bit/44.1kHz audio format too restricting for serious use (in fairness, the freeware version is probably intended as a taster for the $90 professional version).

SEQ24

SEQ24 (http://filter24.org/seq24) should please musicians who already have MIDI synths and want a minimalist sequencer for recording and playing MIDI loops, particularly for live performances, where you don't want to be bogged down with loads of features. It runs under both GNU/Linux and Windows, but Windows users do have to initially install two run-time packages before SEQ24 itself, which might be a little confusing for the novice.

However, once this has been done the application itself is very easy to get into, being similar in concept to hardware sequencers such as the Akai MPC range. The main display contains four rows of eight boxes, each of which can contain a sequenced pattern. When you right-click in a box and select 'new', the Pattern Sequence Editor pops up with a traditional piano-roll editor, where you can enter time signature and bar length (between one and 64) for the sequence, and select the MIDI output port and channel to route it to the appropriate synth. You can play notes in from a MIDI keyboard or draw them in using the mouse, and there are various basic editing tools, including undo, quantise and transpose. There's a snap-to-grid function, and beneath the main note display you can view and edit velocity or any other MIDI Controller information. Buttons at the bottom right of the Pattern Sequence Editor control Record arming, MIDI Thru (so you can hear what you're playing before entering record mode), and Sequence to MIDI bus (which toggles playback of your recorded sequence).

Patterns can be saved and loaded individually in Standard MIDI File Format 1, or it's possible to import several MIDI patterns into the main display consecutively, to form a 'screen set' of up to 32 sequences. You can even switch live between up to 32 screen sets (1024 patterns in total).

Once you've recorded and edited a few patterns, you open the Song Editor window, where you can build complex arrangements with them. Each gets its own horizontal track, into which you can drop multiple instances wherever they are required. There are Mute buttons, so you can bring tracks in and out in real time, Left/Right locators for setting up loops, and some basic tools that delete sections, add new sections and copy existing sections of your song.

It's early days for the Windows version (which is still officially in Beta), but I didn't experience any crashes, and after just a few minutes I really got into the real-time approach to song-making — you can leave everything looping while you write new patterns, and then drop these into your composition or jam over the top. There are copious keyboard shortcuts for making changes 'on the fly', and you can even define incoming MIDI events that turn sequences on or off, for Orbital-style live performance mixes.

It's a shame that SEQ24 doesn't include a metronome (although I quickly created a hi-hat part to perform this function), and that it doesn't support soft synths. (Admittedly, the latter isn't its primary function, and you could set up software routing to a soft synth host using a Virtual MIDI cable such as Maple MIDI Tools, downloadable from www.hurchalla.com/MapleMTv356.zip.) However, if you've already got a bunch of MIDI synths and you're just looking for a stripped-down step-sequencing tool for your live performaces, SEQ24 could be just the job!

Luna Free

Luna is described by its developers (www.mutools.com) as an ultra-light application that's a rock-solid musical tool. It supports both audio and MIDI recording and playback. There are two versions: Luna Unlimited (just 29 Euros) and the Luna Free 'lite' version under review here, which is still surprisingly capable. Both versions run on Windows and Mac OS X. The Windows version only supports low-latency ASIO drivers (but if your audio interface lacks these, you could try the freeware ASIO4ALL wrapper from www.asio4all.com).

Once again, there's a familiar arrange page (here named the Composer), consisting of horizontal tracks containing multiple audio or 'MIDI Sequence' parts. There's a useful set of four editing tools (arrow, pencil, eraser and splitter), plus various keyboard-shortcut editing commands, while across the top of the Composer page are (from left to right) a set of menu buttons for File, Edit, and Help functions (containing a manageable total of just 21 options in all), a simple Transport panel with BPM and looping options, and a Part Property panel.

The latter is where you name each part, and route it using the surprisingly versatile features. The novice could simply leave the default routing of all the audio parts direct to the 'Audio Output', but it makes far more sense to instead do it via one of the eight available Racks, which appear in a separate window. Each Rack is a mixer channel with fader, pan, meter, mute, and six slots where you can insert a chain of VST plug-ins. MIDI parts can also be routed to a Rack, except that you would instead insert a VST Instrument in the first slot, followed by a chain of effect plug-ins, as required.


Luna Free: Offering both MIDI and audio recording and playback, and a surprising number of options, including sophisticated routing, yet boasting a simple and relatively easy-to-use interface, Mutools' Luna Free is a capable and versatile sequencer.

But there's far more on offer for those who want to explore further. Each separate Audio or MIDI Sequence part in a track can be routed to a different Rack with different effects or synths, so you could change a track's treatment part-way through a song (the only other sequencer I know that offers part-based effect functions is Samplitude).

You can also route multiple audio and MIDI parts to the same Rack, to treat them with the same plug-in chain, or route parts to any active slot in a Rack. So if, for instance, you had an effect chain comprising chorus, EQ and compressor, you could route some parts to pass through all three effects while others were simply compressed, or EQ'd and compressed. You can also insert a send from one Rack to another (perhaps to add global effects such as reverb).

Recording is a little unusual, in that you first draw in a part of the required length using the pencil tool, choose between an Audio or MIDI Sequence part and then select a file name using a custom file-selector dialogue that I found difficult to get used to. Audio editing is also basic: you can define a new start point, and there are normalise, gain and mute functions that operate on any selection, but you'll need an external editor for more detailed work.

However, MIDI editing provides more possibilities, with both event-list and piano-roll editors available, the latter with a lower area for editing velocity or other controller data, plus a more general set of sequence tools and various essential functions including quantise, transpose, modify velocity/lengths, legato, and so on. More experienced users can also launch a Modular Plug Area where you can directly edit the connections between plug-ins and synths in your Racks.

Overall, Luna Free is the most versatile of the three sequencers under scrutiny here, offering both audio and MIDI support, plus routing possibilities that occasionally rival those of the shareware powerhouse Reaper (see review in SOS June 2007). It will therefore take a little longer to find your way around. However, it still provides a rather more straightforward interface than those of many mainstream packages, that is also considerably easier to get to grips with.

Trying Out Trackers

If you like building up your songs from step-sequenced samples or VST instruments, you may want to investigate a 'Music Tracker' application. Originally developed for the Commodore Amiga platform, there are now quite a few available for the PC (see my July 2004 PC freeware round-up for more info on Trackers (www.soundonsound.com/sos/jul04/articles/pcmusician.htm). Here are some of the latest and greatest that follow in this tradition:

• According to its developers, Buzz (www.buzzmachines.com) is not a sequencer, nor a soft synth, nor a tracker, but all these things and more. You can route its Machines (Buzz objects which either create or modify sound — there are already over 100 available) in real time in the Machine Editor, which is a free-form graphic patchbay where you can connect together synths and effect chains. You can then create songs by chaining together Patterns in the Sequence Editor. Although Buzz itself is no longer being further developed, there's still a thriving community of Buzz users and Machine creators.
• Psycle (http://psycle.pastnotecut.org) is a 'modular music creation studio' that supports VST instruments and effects in both its own and VST formats, and has various similarities to Buzz, offering an advanced Machine View where you connect virtual components together. It features a 64-track step sequencer with loads of facilities.
• Skale Tracker (www.skale.org) supports VST Instruments, MIDI In/Out and 256 virtual channels, and seems to have an enthusiastic following and an active forum, despite the most recent Beta release being in 2004 (its developer has changed job and location, and no updates have therefore been written for some time). The web site was still out of action when I wrote this feature, but nevertheless you can still download the application itself.

Wavosaur

Many freeware sequencers (and even some commercial ones) require an external audio application for more detailed editing. Most people looking for a free PC audio editor have in the past opted for Audacity (which we reviewed in our PC music freeware round-up in SOS July 2004). This package is currently up to version 1.3.3 and now features new repair and EQ effects, timer recording, automatic project save/recover, to help you avoid losing your precious work after a power cut, and an improved selection bar, although it still manages to keep a modest 2.6MB download size (http://audacity.sourceforge.net).
However, Audacity has recently encountered some competition from Wavosaur (www.wavosaur.com) which, as its name suggests, is a Wave editor, although it also supports multi-channel WAV files, AIFF, Amiga IFF, AU, SND, VOX, VOC, OGG, MP3 and RAW sample formats. At just 171KB, it's also one of the tiniest downloads I've ever come across.


Wavosaur: Many freeware (and even commercial) sequencers rely on an external audio application to perform really detailed edits. The freeware Wavosaur editor may provide all the features you need to supplement your audio sequencing program.

I found Wavosaur very easy to get to grips with, since it uses standard Windows shortcuts for most editing functions, and I liked its ability to use the mouse scroll-wheel for horizontal/vertical zooming of waveforms. You can work with multiple files open simultaneously and copy and paste between them, and there's also a good selection of basic edit functions.

A useful selection of basic audio-processing options, including reverse, bit-reduction, pitch-shifting, fade-in/-out and normalise, is also provided, plus some more unusual effects, such as Truncate (which discards all samples whose level lies above a user-defined threshold) and Gapper (which creates periodic gaps at a user-defined frequency).

Fortunately, Wavosaur does support ASIO as well as Windows WDM drivers, so you get low-latency operation, and it also supports the VST plug-in format, so you can use all your favourite effects. However, like various other aspects of Wavosaur, its VST Rack that accesses these operates in rather an unusual fashion, making it easy to dismiss before you've discovered its true capabilities.

First of all, it doesn't link to a specific VST plug-ins folder: the Load VST button function in the VST Rack window instead lets you browse anywhere on your PC for suitable DLL files. This is flexible, but unnecessarily complicated if you use nested folders for different manufacturers, since you need to navigate to the desired folder. You can chain up to 256 plug-ins, and view them in any combination (although the software can't deal with generic plug-ins that don't have any graphic interface of their own), but it's not until you spot the innocuous tick-box on the VST Rack toolbar labelled 'Processing' that you realise it's possible to audition effects in real time as well as apply them off-line. (Most other applications instead provide a bypass button to allow you to switch the effects off.)

Similarly, in the Tools menu there's a useful selection of view options, including statistics, normal and 3D spectrum analysis and a sonogram, which all provide static displays for the entire audio file. You might conclude that there are no real-time analysis options, until you discover that the Input and Output 'oscilloscopes' both provide a handy selection of phase-scope, spectrum and scrolling sonogram displays in real time, as well as the more normal waveform option.

In other words, Wavosaur is a far more capable program than it at first appears to be, and after an extended session I was impressed, especially as this is an application that's still in its early days. I have noticed a few people reporting instability problems, but then I've also noticed that some people find Audacity frustrating to use. The beauty of the fact that they're freeware programs, obviously, is that you can download both and see which suits you best.

Published September 2007