Thursday, April 24, 2014
PC NotesTechnique : PC Notes
Intel's latest chipsets are finding their way into new PCs — but do they offer any advantages to the computer musician?
With many new features, Intel's latest 925/915 chipsets could take the world by storm — once their performance improves.
As I write this at the end of June, the first motherboards featuring Intel's new 915P (Grantsdale) and 925X (Alderwood) chipsets are appearing from manufacturers including Abit, Asus, Gigabyte, MSI and Intel themselves. They herald the start of a new era for Intel-based PCs, with loads of new features.
Let's start with the new LGA775 (or Socket T) CPU socket. This is a completely new way of interfacing Pentium 4 processors to the system buss, and in a radical departure from previous designs it places the pins on the socket rather than the processor. Motherboard manufacturers have apparently been complaining that this puts up the cost of their half of the assembly tenfold, although initial concerns about the socket's fragility seem largely unfounded.
The extra 297 pins of LGA775 are mostly to supply power, and Intel have also improved thermal performance so that heat can be removed more efficiently. For musicians this is a possible plus, since it enables the use of a new heatsink and large-bladed fan that can apparently spin at lower speeds, and even stop if temperatures drop sufficiently. Intel have already released a new 3.6GHz Pentium 4 model in the LGA775 package, along with repackaged versions of various other slower models. All of them use the Prescott core that Intel first introduced in their Pentium 4E processors to manage clock speeds higher than 3.6GHz, but apart from the new packaging nothing else has changed internally. As with the Dothan processors discussed in last month's column, the new Pentium 4 models adopt a new and confusing numbering scheme, with the 2.8GHz version becoming the Pentium 4 520, while the the 3GHz, 3.2GHz, 3.4GHz and 3.6GHz models follow on as the 530, 540, 550 and 560 respectively.
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More Chips Please
The 925X chipset roughly replaces the existing 875P Canterwood, while the 915P replaces the 865P Springdale range. Both support the new 800MHz LGA775 processors, dual-channel DDR2 memory, up to four PCI Express x1 slots, and one PCI Express x16 slot for the graphics card. DDR2 memory is more flexible than its predecessor, allowing (for instance) a pair of 256MB DIMMs to be paired with a single 512MB DIMM to run in interleaved dual-channel mode, and the memory controller supports a bandwidth of up to 8.5GB/second.
Only the 925X supports ECC (Error Correction Code) RAM for extra security, but the 915 chipset supports a slower 533MHz in addition to the 800MHz System buss, and slower (and cheaper) DDR400/333 memory as well as the faster DDR2 533/400 memory. In practice, most motherboard manufacturers will only provide DDR2 support with the 925X chipset and partner the 915P with DDR, for a significantly cheaper product.
The PCI Express x1 expansion slots use a high-speed serial buss and are only about an inch long, but offer double the bandwidth of a standard PCI socket. For graphics work, multiple 'lanes' are used to increase this bandwidth still further. The 915/925 both support PCI Express x16, using 16 lanes for up to 4GB/second bandwidth to and from the graphics card (ideal for Microsoft's forthcoming Longhorn operating system). Both chips still support up to six PCI slots, but AGP support has been abandoned.
Four SATA-150 ports and one PATA/100 IDE controller are available for attaching drives, and there's Intel's Matrix Storage Technology (MST), which supports two new features. One is NCQ (Native Command Queuing), which executes hard-drive commands out of order if this results in fewer disk rotations or seeks to find particular data, but it's only currently supported by Seagate's Barracuda 7200.7 series. MST also lets you use RAID 0 and RAID 1 on the same set of drives — so, for example, in a twin 200GB-drive setup you could create a 100GB RAID 0 (Stripe) array to achieve more simultaneous 24-bit/96kHz audio tracks, and devote the remaining space to a RAID 1 (Mirror) volume for extra security of other data.
Intel's Wireless Technology is also built into both chipsets, but sadly no-one has yet been able to test this aspect of them, because Intel don't yet have the feature working. Finally, Intel's new HDA (High Definition Audio) replaces the now-archaic AC97 standard with support for up to eight channels of audio in formats of up to 24-bit/192kHz on the motherboard. However, before you get too excited about not ever having to buy a soundcard again, it's best to bear in mind that a typical HDA motherboard codec will only offer A-D (recording) conversion with an 85dB signal-to-noise ratio. Most soundcards designed for musicians manage signal-to-noise ratios of at least 100dB, giving less background noise on recordings.
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Since I last looked at version 4.0 of the versatile universal sample converter CDxtract, in SOS October 2002, it's had quite a few incremental updates, improving the various conversions and adding new destination formats on the PC version — such as SampleCell, the new EXS24, Kontakt, Kurzweil, Reason NNXT, NI Battery, and VSampler 2 import. Now version 4.2 adds several significant new options, including NI Kontakt and Recycle import, export of all supported formats to Muon's DS404, a new DropSound utility for playing WAV and AIFF files, and further enhancements to the existing translations.
CDxtract 5 is already in development and promises a completely new interface and several new features when it appears at the end of this year as a free update to registered users. I don't normally advertise products until I've seen them with my own eyes, but this case is an exception, since the developers are actively seeking suggestions for improvements or new features. Check out www.cdxtract.com.
I reviewed some Tracker software in my PC Music Freeware Roundup in SOS July 2004, and now I've discovered yet another contender. Skale Tracker is based on the Fast Tracker 2 environment, runs under Windows 98, 2000, XP and Linux, and goes beyond many other trackers in supporting MIDI, DirectSound and ASIO soundcard drivers, plus VST Instruments and effects. It also includes a sample editor and an attractive graphic mixing console. If you like the traditional tracker environment but want to take your music further, Skale Tracker looks ideal, especially as it's free! Many thanks to SOS reader Laurence Gillian for the tip-off. More details at www.metamacro.com/awezoom/skale/
I reviewed Iomatic's Registry Medic in last month's column, and since then I've found another related utility that I can recommend. Regseeker not only cleans the registry, but also lets you examine such things as startup entries, colour schemes and application histories, as well as providing a set of OS tweaks like those of Microsoft's TweakUI. However, for me its most useful function is 'Find in registry'. This lets you enter a word or phrase and then displays every reference to it in a single list. This is far easier to use than Regedit's simple Find function (which displays one reference at a time), and is ideal if you're trying to remove references to an old soundcard.
After I had removed Audiotrack's Maya 44 and its drivers from my review partition, Regseeker still found 62 'Maya' matches, most of which had not been added by the soundcard itself, but were references to it in Cubase and Pro-53's lists of ASIO, DirectX and MIDI driver entries. Overall, I didn't find Regseeker quite as thorough as Registry Medic at cleaning the registry, but every cleaner I've tried seems to find a slightly different list of problem entries, so it complements the latter very well. The best news is that it's freeware. Find it at www.hoverdesk.net.
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What Does It All Mean?
So what are the implications of all this for the PC musician? Well, as I said way back in PC Notes November 2002 (when first discussing PCI Express), we won't have to abandon our PCI soundcards just yet. Most motherboards featuring the two new chipsets generally have four PCI slots and two PCI Express ones, or just two PCI slots and one PCI Express slot, plus the new PCI Express 16x graphics slot in both cases. This is understandable, since there are so few PCI Express cards yet available.
The main fly in the ointment is that initial test results with the first 915/925 motherboards are disappointing. Their performance is equalled in many cases by an Intel 875 and P4 Northwood, and they're soundly trounced by AMD's new Athlon 64 3500+, 3700+, and 3800+ models. The 925X does out-perform the 915P by some one to three percent because of its DDR2 RAM, but this option is considerably more expensive considering the small increase in performance. A problem with the Grantsdale chipset also resulted in Intel recalling it just a week after its launch. Moreover, the Prescott core also runs much hotter than the Pentium 4 Northwood series, so for the time being, at least, if you want a quiet PC for audio work, a 915/925 model won't be suitable. As always, it pays to stay on the sidelines until the teething troubles are thrashed out. I suggest at least six months.
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Changing Drive Letters
Do you find that your various drives and partitions seem to be assigned drive letters in an almost random order, so that you're often left searching up and down the list for a particular drive in the left pane of Explorer? This can become particularly annoying with a multi-boot PC, since you may find that particular partitions get a different drive letter in each boot, so they change order in Explorer's list. And have you ever thought how convenient it would be to always have your CD drive appear in Explorer below your various hard drive partitions, rather than wherever Windows thinks fit to place it?
pcnotes figure 2
Finding your data could be even easier once you abandon Windows' drive letter allocation and choose your own, as shown here.
Well, you can do all this, courtesy of the Disk Management extension of Windows XP's Computer Management tool.
Launch the Computer Management tool. Do this from Control Panel, by clicking on the Administrative Tools applet and then on Computer Management, or by launching Windows XP's Run dialogue, using the Windows-R shortcut, and typing 'Compmgmt.msc' (but without the quotes) into its text box and pressing Return.
Next, click on Disk Management in the Storage section, and by default all your drives will appear in the right-hand pane as both a list of volumes and a graphical view beneath.
To change the volume letter (and hence its position in Explorer's list) simply right-click on it in either view and choose 'Change Drive Letter and Paths...', click on the Change button in the window that appears, and then choose another currently unassigned drive letter from the drop-down box.
If you want to swap the letter with another existing volume, just assign it another temporary unused letter, change the other drive to the now released letter, and then change the letter of the first to your desired letter.
You can't change the drive letter for your system or boot volume, as this must always remain as 'C:'. In addition, 'A:' and 'B:' are reserved for floppy drives, but otherwise the sky's the limit. I find themed letters are best, such as 'P' for Projects, 'S' for Samples, and 'U' for Updates. My CD drive is 'Z:', so it always appears at the bottom of the list, and if you have any removable drives, such as external Firewire or USB models or memory cards, these are easier to spot near the bottom too.
You won't be able to change the drive letter on a partition if it houses a page file, although of course you could change the location of the page file first. Also, watch out for any applications that rely on specific drive letters and paths to find associated files — while it's normally fairly easy to relocate audio files belonging to a particular song if they've moved, some apps — such as the Spectrasonics soft synths — lose track of their huge data files if you change the drive letter of the partition involved, and show an error message when you next try to use them.
However, Spectrasonics have now provided the SpectraMove utility on their web site (www.spectrasonics.net) to take care of problems like this for PC user
Wednesday, April 23, 2014
PC MusicianTechnique : PC Musician
Shareware might be cheap, but it can also make you very cheerful. We round up some of the best on the current PC music shareware scene.
You can't always judge a book by its cover, and this is definitely true of Audiomulch, whose 'interactive musician's environment' is incredibly flexible and user-friendly for anyone who wants to create new sounds.
Back in SOS July 2004 I rounded up some of the best freeware available for the PC musician, but mentioned that if you were prepared to indulge your credit card a little there was lots more low-cost software available as shareware. The difference between freeware and shareware is fairly obvious — developers want you to pay for the latter — but the difference between shareware and full commercial software releases is becoming increasingly blurred.
Strictly speaking, shareware is copyrighted software that you can try before you part with any money and, unlike the demo versions available from commercial developers, you can normally download the full version to try out. Some shareware authors let you download the full version with no restrictions and trust you to be honest enough to make a small donation if you carry on using the software. However, most let it run for 30 days before timing out, or provide a slightly reduced feature set, occasional bursts of low-level noise, 8-bit rather than 16-bit or 24-bit support, or add increasingly frustrating nag screens to encourage you to do the right thing.
Whatever the approach, once any trial period is over the author expects you to make a donation or pay a registration fee to carry on using the software, although this fee is generally a lot lower than the cost of commercial releases because shareware authors' overheads are much lower — generally no commercial premises need to be rented or teams of staff paid, for instance. In return for your registration, you will normally get valuable technical support by email, extra documentation, some free updates in the future, and often a registration number that uniquely unlocks extra features in the software (or removes any trial limitations such as those just mentioned). There's a lot of PC music shareware out there and this month I'm going to round up some that I've found to be most useful.
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Most of the advice I gave in SOS July 2004 about where to look for freeware applies equally well to shareware. Indeed, there seem to be very few web sites that specifically cater only for the latter. Of course, once you open out your search from music-related software to PC shareware in general there are lots of new sites worth exploring.
CNet's Download.com (www.download.com) is one of the most obvious, with 12 download categories, while WindowsPC (www.windowspc.com) provides details of PC shareware and freeware in 26 categories, including Multimedia Tools, which itself includes 191 items spread over 92 pages. SourceForge.net (http://sourceforge.net) is the world's largest Open Source software development web site, and if you click on their software map link you'll find tens of thousands of projects in 18 categories. However, they are well organised into sub-folders so it's easier to find items of interest than you might expect.
The main problem with such huge sites is tracking down the most appropriate shareware for your purpose, especially when there may potentially be dozens of software items with a similar goal. I do my best to alert SOS readers to any of particular merit and some sites, such as Download.com, do also provide occasional reviews, but for my money the most helpful site is ZD Net (www.zdnet.com), whose downloads area not only hosts 14 categories (each with further sub-divisions), but also features lists of the most popular downloads and the most talked-about downloads, and offers product spotlights, as well as letting you sort site contents by Name, Date Added (if you want to rule out old Windows 95 utilities, for instance), User Rating (to discover what other people think), and number of Downloads (their ultimate vote).
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Audiomulch already has a cult following, includes among its users Kieran Hebden (aka Four Tet — you can read our interview with him in SOS July 2003), and is one of the only PC-only applications I've come across, apart from Gigastudio, that has Mac users drooling.
So what's all the fuss about? Well, Audiomulch is described as an 'interactive musician's environment', will run on nearly all versions of Windows including 96, 98, ME, NT 4.0, 2000 and XP, and supports soundcards with ASIO, MME-WDM and DirectSound drivers. The version I downloaded was 0.9b16 (it's still officially a Beta release), although Australian Ross Bencina has been developing Audiomulch for a total of five years. Each released version is a free download and not restricted in any way, but times out on a certain date, encouraging you to download a more recent one. However, if you register, for $50, the shareware warning messages disappear, versions no longer expire, and you are eligible to download all future versions up to (but not including) 2.0.
With Audiomulch you can either treat sounds coming through the input of your soundcard, treat an existing audio file, or generate audio files from within the application. You then pass them through a network of 'Contraptions' (what a lovely choice of term) that constitute a 'Patch', strung together using virtual cables, before sending them to the output of your soundcard or exporting them as a WAV or AIFF file with a bit depth of up to 24-bit and sample rate up to 192kHz.
Despite its toy-like appearance, Fractal Tune Smithy is a surprisingly versatile MIDI utility to suit both the dabbler and the academic. It can help you incorporate more exotic instruments and tunings into your music, as well as generating plenty of new ideas.
Audiomulch is so simple to use that you almost don't need a manual, although the detailed help file is well-written and informative. Essentially, the user interface is divided into three 'panes'. At top left is the Patcher pane where you use a right-click menu to select your chosen Contraptions, and a left-click to select them, drag them about, and attach patch cords to string them together. Most Contraptions have a 'Property Editor' (essentially a front panel featuring knobs and sliders) that can be opened in the Properties pane at top right. The third and final pane is for Automation, where you can alter any front-panel parameter over time. Each parameter you decide to automate appears in this pane as a graphic vector lane. You can click anywhere on its timeline to add further points to it and drag them backwards and forwards, up or down.
Apart from the automation, we could still, so far, be talking about DIY synth/effect designers such as NI's Reaktor or Tassman from AAS, but the huge difference is that Audiomulch is optimised for live performance. You can get flexible and rewarding results with just a handful of Contraptions, but, best of all, you can re-patch them or introduce new ones and connect them to the existing setup during playback without skipping a beat or disturbing the flow of audio in any way. For this reason, Audiomulch is also perfect for sound designers, since it makes it possible for them to generate and capture continuous, evolving soundscapes and snip the best bits out afterwards — just the thing when inspiration strikes!
The Contraptions themselves fall into seven categories: Input/Output, Signal Generators, Effects, Filters, Busses, Mixers, and Beta (new and experimental ones that may not yet be completely bug-free).
Input/Output: This section is for patching in your soundcard's in and out, and caters for multi-channel devices via its auxiliary ins and outs.
Signal Generators: This is where things start to get more interesting. There's the '10 Harmonics' additive oscillator, a comprehensive dual-oscilllator arpeggiator, a monophonic Bassline synth, Bubbleblower sample granular synth, a sample-based drum machine, looping Fileplayer for replaying existing audio files, Loop Player for doing the same thing but synchronised to MIDI clock, the weird, infinitely-sliding Risset Tones generator, and the simple sine/noise Test Gen.
Effects currently comprise the bit-depth and decimation options of Digigrunge, the DL Granulator delay-line granulator, a clutch of more traditional Flanger, Nasty Reverb, Phaser, harmonic Shaper, Ring AM and stereo delay options, plus the more intriguing Pulse Comb (a sort of delay line where each repeat has its own envelope), and my favourite — SSpat, a stereo spatialiser that allows you to set the path and trajectory of a moving input signal.
Filters: There are only five of these, including traditional mono and stereo parametric EQs, but on offer are some interesting options, including 5 Combs, which adds resonating chordal drones; Risset Filters, which provides a bank of up to 50 moving band-pass filters for an intriguing range of possibilities; and Nebuliser, another granular synthesis variant that passes each grain through a band-pass filter.
Busses: This section contains Contraptions that conveniently combine up to 12 mono or stereo signals, although I also found I could patch multiple signals directly into any Contraption's input or output, which makes it easy to patch as you go without interrupting an existing signal path.
Mixers: The selection here offers various permutations of mono and stereo mixer with added level and pan controls, plus a selection of Matrix Contraptions for dynamically switching routing between various input sources and output destinations, crossfaders to move smoothly between them, and Inverters and Gain controls.
Beta: Amongst the current offerings here is a useful Frequency Shifter, the currently semi-impenetrable 16-voice Cannon Looper, the 16-track Live Looper, and the many permutations and complexities of South Pole, a four-pole resonant filter with multiple envelope followers and LFOs.
VST plug-ins and instruments are also supported, so you can patch any of your existing collection into an Audiomulch design. A new Contraption will appear with the appropriate number of inputs and outputs, and you can launch the plug-in or instrument's normal interface in Audiomulch's Property Editor pane. You can also sync Audiomulch to an external MIDI sequencer, and allocate Contraption parameters to MIDI controllers.
So you really do get the best of all worlds — the free-form 'patchability' of the Audiomulch interface, the ability to use within it any combination of its own Contraptions and your existing VST collection, and the added flexibility of running it alongside your existing MIDI + Audio sequencer. The graphic interface won't win any awards for its beauty, but is extremely functional and easy-to-use. The program also has never crashed on me.
Compared with a standard MIDI + Audio sequencer or pre-mastering application such as Wavelab, some of the features I found invaluable were the ability to set up several parallel chains of effects on one source, for more complex treatments, and the ability to add several generators at different points in the chain, introducing subtle background sounds to complement the main one.
With sufficient thought and some lateral thinking you could probably use your favourite sequencer to set up some of the routings I tried, and automate some of their parameters in similar ways, but with Audiomulch it's just so much easier and such a lot more fun — plus you never have to stop playback while you patch in further Contraptions and alter routings. The word that kept popping into my head all the time I was using Audiomulch was 'freedom'.
I found this program great for creating evolving soundscapes, background drones and ambiences, and it's also one of the cheapest ways to explore granular synthesis. Anyone who creates new sounds destined for sample CDs should definitely have this in their software collection. Audiomulch is also being used for avant-garde and experimental composition, and live performances of loop-based techno and trance, either as a self-contained environment or in conjunction with other software. No wonder it's gained a cult following. I'm taking out full membership immediately!
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Fractal Tune Smithy
Ever stuck for inspiration? There have been various MIDI-based music-generating applications launched over the years, and many MIDI + Audio sequencers also have integral tools for modifying MIDI data, such as the old Interactive Phrase Synthesiser of Cubase VST. Fractal Tune Smithy (www.tunesmithy.co.uk) is a music generating program that runs both as a stand-alone utility on any version of Windows from 95/NT onwards (including XP), and as an MFX plug-in. According to its advertising material, it "plays tunes as intricate as snowflakes", which is certainly an intriguing prospect — but FTS goes a lot further than generating random sequences of MIDI notes. The most important additional feature is its ability to retune any MIDI synth to a completely new scale tuning. It does this using standard pitch-bend data to create notes that sit 'between' those of normal equal-tempered tuning.
There's a huge number of scales to choose from. The default list contains 32 options, including standard 12-tone equal temperament, plus 15-tone to 31-tone equal tempered, just temperament, and various ethnic and folk tunings, such as Gamelan from Indonesia and Java, Japanese Koto, West African Xylophone and Indian Raga. Further drop-down lists cover historical and modern twelve-tone scales, bagpipes, Idiophones, and even wind chimes. So if you want to play a keyboard in the authentic tuning that Bach or Mozart would have used, you can.
Anyone with a convolution reverb that accepts WAV impulse responses should investigate Voxengo's Impulse Modeler if they want to design their own acoustic spaces.
The handy Play button above the drop-down Scale menu auditions an octave's worth of your selection, and you can also play in real time from your keyboard in both monophonic and polyphonic modes (this is termed MIDI Relaying). The MFX version lets Sonar and Cubase SX users patch FTS in as a real-time effect; you could use MIDI Yoke or a similar utility to do the same in other sequencers.
Various note mappings are used to make it easier to play unusual scales from a normal keyboard. For example, the Japanese koto has just six notes per octave, and these have been mapped onto the white notes, with the black notes generally identical in tuning to the next highest white note. Within a few seconds I was playing what sounded (to me!) like authentic Japanese music.
Of course, it helps if your MIDI synth is playing a suitable sound, and while FTS can choose these for you using General MIDI, you'll get much better results by skipping these automatic choices and allocating higher quality voices by hand at the synth or sampler end. Developer Robert Inventor (aka Robert Walker) has obviously used FTS with more up-market gear, as his web site provides details on how to use it in conjunction with Gigastudio. MIDI merge from multiple input devices is also supported, as are multiple output devices so that you can route different FTS channels to different synths or samplers.
Since music makes more sense if you stick to recognised scales, arpeggios or modes, along with the desired tuning, the drop-down Arpeggio box lets you choose from shedloads of options (a couple of hundred, at least). Along with the better-known options, such as Major, Minor, Diminished, Whole Tone and Diminished Seventh, there are many more exotic modes containing two, four, five, six, seven, or eight or more notes per octave. To give you an idea of the available scope, a few examples selected at random are Messiaen Truncated Mode 5, Bi Yu (China), a set of Raga options, and 'Half-diminished Bebop'!
The next stage in the FTS process is to choose a Musical Seed — a sequence of intervals, plus a note duration, which provide the starting point for generating tunes. There are plenty of examples available and you can even create your own in a variety of ways, from typing in a set of numbers with spaces between them, to playing them in as notes from a MIDI keyboard, or using the mouse and PC keyboard.
The final stage in the journey is the Fractal Tune — a combination of Scale, Arpeggio and Musical Seed, plus a choice of one or more instruments across multiple MIDI channels. Using the Play button now generates continuous, ever-changing tunes based on your settings. If you want impressionistic flurries and cascades of notes, try 'rushes blown in a storm' or 'echo effects in rests'. More extreme examples include 'Fibonacci rain shower', the unsettling 'Paleolithic field recording', and 'bird calls with Afro-Caribbean percussion'.
Don't go away thinking FTS can only generate avant-garde meanderings for classical and jazz buffs. Although many of the offerings are 'off the beaten track', they may still inspire new songs, while others, such as 'string quintet' and 'shakuhachi and koto' are gently melodic, and still others (such as 'resting in the shade') create floating backdrops. You can also explore the more rhythmically-based offerings, such as the improvised 'percussion medley' and 'non-repeating bongos'.
The best way for any new user to get creative is to take advantage of the 'Randomise Now' key, which randomly chooses a new set of parameters each time you press it. Once you hear something that takes your fancy you can start fine-tuning it in every sense, and save the honed and polished version as a Tune Smithy file. You can also capture the FTS MIDI output to incorporate in your work, or record its audio output as a WAV or other audio-format file, It's even possible to make a musical 'e-card' with embedded MIDI data, to send as an email.
Even after quite a few hours delving into this Tardis of a utility, I felt there was still lots more to explore, and while this does make Fractal Tune Smithy rather overwhelming at first sight, you're unlikely to get bored with it. The HTML help-file is a mine of information about the various features, scales and arpeggios, and there are also 75 'Tips of the Day' that point out things you may not have discovered for yourself.
Whether you want to while away an afternoon generating new ideas, modify existing MIDI files, attempt to play exotic instruments with unusual tunings and scales, or explore the more academic and mathematically-based composition techniques, Fractal Tune Smithy can help. While its colourful graphics do make it look initially like a toy, this is a serious tool that's capable of a wide range of musically useful results if you're prepared to spend some time with it. It does support skins for those who prefer a more subdued appearance, but once I'd heard what it could do I didn't really mind what it looked like. Just download the 3MB demo file, complete with 1MB of help, tutorials, and FAQs, and decide for yourself. Depending on what features you want to use, there are various payment levels, ranging from $14 to $45.
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It may not look very exciting, but Voxengo's deceptively simple Deconvolver utility lets you capture the sounds of real acoustic spaces (as well as of audio hardware and software) for use with playback-only convolution reverb plug-ins.
If virtual acoustic spaces don't float your boat, you may be more interested in capturing the sound of a real one, or that of some existing hardware gear or software plug-in such as a reverb (including, dare I mention it, other convolution reverbs), or indeed any effect, preamp, guitar amp, amp simulator or other device that you can pass a test signal through.
Voxengo's Deconvolver is a simple utility with a tiny $23 price tag. It can generate suitable sweep-tone source signals that you send to the input of your test device (or to a loudspeaker in the environment you want to capture). You then record the output from the device (or the sound of the space via a microphone) using any software application or hardware recorder. Those involved with film or other location work could use these tones to quickly capture live acoustics for post-production use.
Deconvolver then provides an easy way to recover the impulse response from these field or device recordings, and accepts them in a wide range of input sample formats. After calculating the impulse responses, it normalises them to a peak level of -0.4dB and can then write them in a wide range of formats, from 8-bit to 32-bit and any sample rates, ready for use in any convolution reverb that accepts WAV files.
A Minimum Phase transform option is provided that sometimes creates better-sounding impulses of devices such as loudspeakers and amplifiers (you may otherwise get some pre-echoing), and there's also a Reversed test tone option that can work better with noisy acoustic recordings and hardware with a limited bandwidth. Since most experimenters end up with lots of recordings made in the field, Deconvolver also provides batch processing to convert them all in one go.
There are a few caveats to getting good impulse responses — such as making sure there is some silence at the beginning and end of the original recordings — and you'll probably end up creating a few before you get the hang of things, but I found that Deconvolver worked very well indeed. For the best results you should top and tail any silences from the final responses, although Christian Andersch of Noisetime has even developed a handy stand-alone batch-file utility that will do this for you automatically (www.noisetime.com/silrem.html), as well as normalising files to any peak value.
By the way, anyone interested in buying Voxengo's Pristine Space would do well to consider the Convolution Suite, which bundles Impulse Modeler and Deconvolver for a total price of $159, a saving of about $43.
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Voxengo Impulse Modeler
I make no apologies for including a convolution reverb section in this shareware roundup as well as in my recent freeware one. A huge number of PC musicians are investigating or already using these plug-ins to provide realistic acoustic spaces.
On the shareware scene the most popular convolution reverb playback plug-in must be Voxengo's Pristine Space (www.voxengo.com). John Walden gave this a thumbs up in last month's SOS review, but a few days after that issue went to press a newer version (1.2) was released. New features include support for multi-channel and AIFF impulse files, faster loading and recalculations, automatic resampling if any loaded impulse differs in sample rate from that of the host application, new search path settings to locate impulse files that have been moved on your hard drive, and some tidying up of channel and slot options. Like John, I'm impressed with Pristine Space, even when compared with the Waves IR1 running the same impulse responses.
However, unlike IR1, which comes with a world-class two-CD IR library, Pristine Space doesn't have a bundled library of its own, although Voxengo do have two other plug-ins in their range that help considerably in this respect. One of their very first releases was Impulse Modeler, which I briefly mentioned way back in PC Notes October 2002. At the time, its ability to design acoustic spaces using a simple graphic plan view and then generate an impulse response from them was useful for users of Sony's (then Sonic Foundry's) Acoustic Mirror and Samplitude's Room Simulator. Now its potential user base has also expanded to PC users of SIR, Pristine Space, Waves IR1, and indeed any convolution reverb plug-in that accepts impulse responses in WAV format.
The latest version of Impulse Modeler (1.7) is also quite a bit more sophisticated, although the principles remain the same. First you define the end of a solid wall by dropping a 'vertex', then you drop another elsewhere to define the other end, and finally you choose a material — a pre-defined selection of these is already available and includes carpet, concrete, drapery, glass, gypsum board, plaster on lath, and plywood panelling, or you can create your own.
You can drag the vertices around to extend or move your walls. By default, the workspace is 5x5 metres, but a Scale slider lets you immediately make the whole design bigger by up to a factor of fifteen, or smaller right down to a hundreth of its nominal size. The surrounding air or other medium can also be defined — so you could, for example, simulate underwater acoustics.
Once your floor design is complete, you then add one or more 'emitters' with variable direction and dispersion characteristics to act as sound sources (you can think of them as loudspeakers), and 'Recepting' walls that act like microphones to pick up the sound of your 'room'. Finally, you can generate your impulse file in WAV format at a sample rate of anywhere between 44.1 and 192kHz, and a bit depth of between 8 and 32 bits. Impulse Modeler calculates all the sound reflections until they decay to a very low level, which, depending on the complexity of your design, may take quite a few seconds, or even minutes.
Admittedly, Impulse Modeler only works from a two-dimensional ground plan, but even so the results can be as dense and believable as a real three-dimensional acoustic space. You can stick to traditional structures such as rooms, halls, theatres, and churches, but the beauty of Impulse Modeler is that you can also investigate other-worldly structures, so if you want to hear the reverb characteristics of rooms shaped like an egg, seashell, triangle, circle, coiled labyrinth or irregular cave, this is the tool for you.
Despite a simple and workmanlike interface, Impulse Modeler is a serious artistic tool capable of a wide range of usable reverbs, echos and delay effects that really do sound good with any compatible convolution plug-in. It should not only appeal to game and film sound designers who need to realise the sound of an unusual structure, but also to anyone who's interested in space — the final frontier. For just $39.95, it's also extremely good value for money.
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Exact Audio Copy
Strictly speaking, Exact Audio Copy (www.ExactAudioCopy.de) is Cardware, since although its German author, Andre Wiethoff, does welcome any donations you care to send via PayPal, he only specifically requests registration from his users in the form of a nice picture postcard showing your home town or its surrounding countryside. However, his utility performs an extremely handy function for musicians: ripping data accurately from Red Book Audio CDs.
If you want to extract CD audio tracks accurately, Exact Audio Copy will live up to its name where many others fall by the wayside.
Before you skip this section because your audio editor already provides a DAE (Digital Audio Extraction) function, hang on a minute, because Exact Audio Copy was written out of frustration with many such functions that sometimes leave your 'rips' with rhythmic clicks and pops, due to scratches or other CD surface blemishes. You probably won't hear these when playing the same CD in your hi-fi, because the hi-fi employs error correction to mask all but the most severe read errors (which mute the player's output momentarily). CD-ROM drives don't implement such error correction during DAE.
Audio data is digitally extracted in chunks, and the start and end points of these are matched up to create a continuous file. Most DAE functions in audio applications extract extra data so that the chunks overlap. They can then be exactly positioned end to end so that the overlapping portions are identical. This is termed re-syncing or jitter correction, but it still doesn't guarantee that the extracted file is totally a bit-for-bit copy of the original. To achieve this you ideally need a CD player and PC application that both understand 'C2 error flagging'. Your software is then aware of any read errors and can re-read the sector in question.
Exact Audio Copy offers a speedy 'Burst' extraction mode with no error correction, and a 'Fast' mode with jitter correction similar to those provided by most other applications. However, its main strength is 'Secure' mode, which can use C2 error flagging if your drive supports it (although many don't), but more often relies on its own multiple extractions of the same sector, re-reading and comparing until it's sure that the data has been read perfectly.
While the aforementioned process can make it much slower than other copiers, especially when reading a scratched or otherwise damaged CD, the digitally extracted data will be as near 'perfect' as possible. You also won't have to listen through each track to make sure it's fault-free, since EAC also displays an informative 'Status and Error Messages' window after extraction has finished. This provides details of track quality (anything less than a 100 percent report means that bad sectors were found but EAC's secure mode corrected them after a re-read), plus any uncorrectable read errors despite multiple reads.
During my tests with EAC I became increasingly aware that many CD tracks (even those with a reported 99.9 percent track quality) would have slipped through most DAE functions with read errors, while other extractions of tracks with bad scratches might not have been usable at all with lesser utilities. Although EAC supports CD Text and database file naming, saving of tracks with a wide variety of audio compression formats and a host of more advanced functions, it's the 'secure copy' function that will be most prized by the majority of musicians. If you want an exact audio copy, you now know how to do it.
Tuesday, April 22, 2014
PC MusicianTechnique : PC Musician
Extra DSP assistance to help your PC's processor cope with effects treatments used to be the province of the pro. Now there's a wide range of DSP-equipped cards to fit all budgets — but many people don't realise the latency issues that might be involved in using some of them.
Last month's PC Musician dealt with how you can add DSP effects to your PC setup by plumbing external processors, such as reverbs, into your soundcard and sequencing software. This idea is a useful way of stretching the power and facilities of your system, but if it's not something you want to do, there are still ways of taking the stress off your PC processor and giving yourself more effects power at the same time. A surprising variety of DSP-assisted audio effects are available for the PC musician, ranging from soundcards with on-board effects to dedicated DSP expansion cards that provide no audio I/O, instead concentrating on a range of high-quality effects and relying on a normal soundcard, installed alongside it, to get audio signals in and out of the computer. Strictly speaking, all software plug-ins use Digital Signal Processing, but to avoid confusion let's refer here to DSP as anything using dedicated hardware DSP chips separate from the PC's main processor.
Soundcards + DSP
Soundcards that have DSP effects include budget models such as those in the SoundBlaster Audigy range, whose algorithms are now quite sophisticated, and Yamaha's incredibly successful SW1000XG, with its five effect busses.
Yamaha's 01x turns the concept of extra DSP for your PC into a deluxe studio centrepiece.
Staying with Yamaha for the moment, their DSP Factory has a 16-channel mixer with 4-band EQ and a dynamics processor on each channel, plus two multi-effect processors. When these cards were released in 1998 they were little short of revolutionary, but unfortunately their lowest possible latency, of 23ms, isn't up to (or rather down to) today's standards. However, I've noticed some enterprising musicians using a second basic stereo soundcard for low-latency recording, but switching back to their Yamaha products during the mixing process.
Those looking for quality reverbs were also tempted in 1998 by Lexicon's Studio 12T system, effectively a soundcard with the engine of the famous PCM90 hardware reverb grafted on to it. Unfortunately, this proved tricky to set up in a suitable PC, and while its £2600 price tag was probably in line with hardware units of similar quality, relatively few musicians were tempted. The subsequent Lexicon Core 2 released in late 1999 was far more affordable, at £599, but still wasn't easy to configure and didn't set the world alight.
Bringing the options right up to date are products like Yamaha's new 01X (reviewed in SOS March 2004), which not only combines the features of a soundcard with a fixed bank of DSP effects, but also adds a digital mixer and remote-control functions. With eight analogue inputs, a further 16 channels controllable via software, plus dynamics and 4-band EQ on every channel, the 01X integrates neatly with the majority of sequencer applications, to form a poweful studio system.
Going a few years back in time, for those who wanted more flexible DSP assistance 1999 saw the release of the long-awaited Creamware Pulsar system. This soundcard and DSP combination, along with the subsequent and more powerful Scope Fusion platform, has gone from strength to strength, largely due to its array of Analog Devices SHARC DSP chips. These chips provide raw DSP power that can be allocated at the user's whim to whatever combination of audio effects, software synthesizers and samplers can be run within the capability of each card. Those needing more power can add expansion cards containing yet more DSP chips.
While not cheap, Creamware systems provide excellent audio quality, although most users would probably agree that the emphasis has tended to be more on the synths than the effects. Moreover, most of the Creamware cards don't provide on-board RAM, so while their Masterverb Pro reverb seems to be roughly equivalent to a mid-range hardware device, it does put a lot of traffic on the PCI buss as the audio data gets passed to and fro from your audio application, which may restrict the maximum number of audio tracks you can run. Nevertheless, the XTC version of the Pulsar card provides DSP-powered VST effects and Instruments that integrate neatly into applications like Cubase, and may be just the combination you need.
Soundscape's Mixtreme is another popular soundcard/DSP combination originally launched in 1999 that's still going strong (admittedly after suffering from a lot of support uncertainty after various take-overs). Happily, in July 2003 its original developers and support group re-assumed responsibility for Soundscape products, creating a new company called Sydec Audio Engineering (www.sydec.be).
Like Creamware's range, Mixtreme's DSP power is fully user-configurable, but concentrates on mixing and effects, with exemplary audio quality. If you need more DSP power you can run multiple Mixtreme cards on a single IRQ, or add Mixpander cards, and their DSP mixer's 'stream' inserts are available as inputs and outputs to Windows applications, so you can add DirectX or ASIO effects to individual channels as required.
Because of its close connection with the Soundscape Digital Audio Workstation range, Mixtreme has also attracted rather more high-end third-party plug-in effects than Creamware, including a Drawmer compressor, TC Works (now TC Electronic) reverb and multi-band Dynamizer, plus a world-class DeHiss from Cedar.
Hints & Tips For DSP-Assisted Soundcards
Creative's Audigy Platinum eX.
Creative's SB Live!, Audigy, Audigy 2, and Audigy 2 ZS soundcards all provide DSP effects, but only with the Audigy 2 Platinum eX and Audigy 2 ZS Platinum Pro models do you get the option of applying several of these effects in different amounts to individual audio tracks, rather than in a blanket fashion to the entire stereo audio output emerging from the soundcard.
You can access the DSP effects individually when using Creative's low-latency 16-bit/48kHz ASIO drivers (although they are not available if you choose the 24-bit/96kHz ASIO driver option, which would require twice as much DSP power to run the effects). Within ASIO-compatible applications such as Cubase and Logic Audio you'll see two extra stereo output channels labeled 'FX Slot 1/2' and 'FX Slot 3/4' (see screenshot). FX Slot 1 is permanently wired to the Audigy DSP reverb and slot 2 to its chorus, while slots 3 and 4 are whichever two other effects you 'Add' in Creative's EAX Control Panel; you can set these up as multiple send effects from each Cubase track.
All the DSP effects routed in this way will now be heard via the main stereo output of the soundcard, but if you want to record your final stereo mix, complete with these effects, as a new pair of audio tracks in your sequencer application, you can use the Creative ASIO input labelled 'Post EQ Front L/R'. Sonar 3.0 users should be able to access the same features, since this version supports ASIO as well as WDM drivers.
In the case of Yamaha's popular SW1000XG, its six stereo audio playback channels each have access to reverb, chorus and variation busses, so you can apply any amount of these three effects to each channel, and you also have the option of adding your choice of up to two insert effects to any one of the six. This is wonderfully flexible if your songs use up to six stereo tracks, but what if they have a larger number of mixed mono or stereo tracks?
The answer is to use SW1000XG outputs 1/2 for the main audio output; route Aux Send 1 signals to SW1k outputs 3/4 and configure it for 'fully wet' reverb; route Aux Send 2 signals to SW1k outputs 5/6 for 'fully 'wet' chorus; and so on. Just as with the Audigy, you can then apply your chosen amount of any of the five effects to any of your audio tracks, however many you have.
So there are some good systems available — but for many musicians wanting to add the ultimate in DSP-assisted effects to their software application, the two most serious products for the shortlist are TC Electronics' PowerCore and Universal Audio's UAD1. These are cards which provide only the extra DSP muscle for running plug-ins and feature no I/O, so they have to be run in conjunction with a 'normal' soundcard. Both the TC and Universal Audio systems support the use of up to four DSP cards, if your computer has sufficient slots and you want yet more power.
However, it's important to point out that the proprietary 'free-form' DSP solutions mentioned here can only run plug-ins specifically designed for their own hardware — you can't use them to run loads more standard VST or DirectX plug-ins than you would normally be able to run on your PC's native processor.
It's always difficult to compare the power of products featuring different DSP chips, but both cards have an impressive range of plug-ins developed especially for them by a variety of companies. The PowerCore has the reputation of having the best available reverb plug-in, and many musicians are also swayed by the quality of its Sony Oxford EQ, its Master X5 finaliser, the TC-Helicon Voice Modeler, and the Waldorf D-coder vocoder. It's now available as both a PCI card and an external FireWire-connected unit that has double the DSP power of the PCI card. On the PC, Powercore supports the VST plug-in and Instrument standard, so you don't have to employ proprietary routing. The Powercore plug-ins appear as options in your existing VST plug-in list inside any VST-compatible host application.
The Universal UAD1 card has some of the best compression plug-ins that are currently available bundled with it (the Teletronix LA-2A Levelling Amplifier and UA1176LN Limiting Amplifier), and its Pultec Program EQ (EQP-1A) is also well-respected, as is the RealVerb Pro reverb — although it's perhaps not so coveted as TC's bundled reverb. (UAD's new flagship DreamVerb may change this situation, however.) Another tempting UAD1 plug-in is the high-end 'analogue' Cambridge EQ. The latest version 3.3.1 of the UAD software supports both VST and DX plug-in formats, so you can use it from within most audio applications.
Playback & Input Monitoring Latency Scenarios
PCM DSP 1
1. Audio playback with soundcard featuring DSP effects: When audio tracks are being played back, RAM buffers are required between the computer and soundcard to avoid glitching, but passing the audio signal through dedicated DSP effects on the card won't add any extra latency, whether they are fixed in ROM or downloaded from your computer into RAM on the soundcard.
PCM DSP 2
2. Audio playback with dedicated DSP card plus normal soundcard: If you add a dedicated DSP card to provide effects, it will need buffering in exactly the same way as the soundcard, so playback latency will triple. Fortunately, the majority of music applications compensate for this extra path delay automatically, so that all your tracks remain in sync.
PCM DSP 3
3. Zero latency input monitoring with DSP-equipped soundcard only: The simplest path for input monitoring is the so-called 'zero' latency option provided by the soundcard, which connects the A-D to the D-A converter. This results in about 2ms overall latency (one millisecond for the A-D and another for the D-A conversion). The input signal may also pass through the soundcard's effects block with no added latency, as shown here.
PCM DSP 4
4. Input monitoring through sequencer application, with added DSP effects card: The most complex signal path is created by monitoring an input signal through the application, as shown here. Buffering is required for the input when it is sent to the application, as well as for its playback, resulting in double the normal latency (the two upper sets of buffers), although this does allow you to add native plug-in effects. If you want to add dedicated DSP effects on a Powercore or UAD1 card, two further sets of buffers are required, so overall latency will quaduple.
We've established that there's a range of options available for providing more effects and processing muscle — but there are things to bear in mind when deciding to incorporate one of these options into your setup. One potential problem when using DSP effects is a more complex signal path, which may result in audio delays. The simplest configuration is a soundcard with integral DSP effects, such as the ones mentioned above, since these effects can be applied to playback channels while the audio signal is in the soundcard. Because these signals are already in the digital domain, there's virtually no added delay in processing them, so the mechanism of applying the effects is relatively unproblematic for the user.
The situation when adding DSP effects to incoming analogue signals while the signals are actually being recorded is very similar to the above. As the audio has already been converted to digital by the soundcard's A-D converter, digital effects can be added in the digital domain without any further delays being apparent. Most products with DSP processing functions provide options either to listen with effects but only record the dry signal (ideal if you want to monitor vocals with a little added reverb, for instance), or record the signal with the effects. This approach is perhaps more suitable for electric guitar (for instance), although once again most people prefer the dry option, so that they can change their minds about their effects later on.
In short, then, cards such as the Audigy, SW1000XG and DSP Factory, the original Pulsar and the Mixtreme, plus all-in one solutions like the 01X, can all add their effects without you experiencing any delays. Moreover, in the case of the Pulsar, if a soft synth has been selected and loaded into its DSP chips, almost the entire signal chain is running inside them, and if you use its hardware MIDI input to trigger the synth, overall latency will be very low.
Wait For It
In the case of 'freeform DSP' systems such as the Mixtreme, Pulsar XTC, PowerCore and UAD1, operational delays may be experienced when initially sending the desired data to the DSP chips. So, for instance, if you're using the Pulsar XTC from within Cubase, when you choose any Pulsar plug-in from the drop-down list it may take longer to appear on your screen and be usable than a plug-in relying on native processing, since its data must first be loaded from your hard drive, then downloaded to the DSP chips via the PCI buss, and finally initialised.
More problematic are audio signals that have to be ferried to and fro inside the PC. This is when buffering is needed to maintain a glitch-free audio stream against untimely interruptions from other Windows tasks. When you record an input signal (a guitar or voice, say), it's subject to software buffer delays (latency) on the way into your Windows audio application. The playback of existing audio tracks is subject to similar delays before the signal reaches the soundcard and is converted back to analogue through the D-A converters, so that we can listen to it.
When you're using a DSP effects card such as the Pulsar XTC, PowerCore, or UAD1, where the effects aren't applied directly to a soundcard's output but are instead added elsewhere in the signal path, such as to an insert or aux send, extra buffers are needed, and more delays are involved. For example, if you're using Cubase and a soundcard, all playback signals are subject to the normal latency caused by the ASIO buffer size. If you add a UAD1 card, and configure it to provide a VST insert plug-in, the audio is first sent from Cubase to the DSP chips on the UAD1 card, incurring a latency equal to the ASIO buffer size, and then returned to Cubase, incurring a further latency equal to the ASIO buffer size, before being routed to your soundcard for playback, when an additional latency equal to the ASIO buffer size occurs.
So if you were running with a 256-sample buffer at 44.1kHz, playback latency would be about 6ms, but the additional latency due to the UAD1 insert would be 512 samples, or about 12ms, making overall latency 18ms. If you passed your audio through several insert effects, each one would add a further 12ms to the overall latency. (See the diagrams overleaf for an illustration of where latency occurs in the signal path of different PC/soundcard configurations.)
Many modern audio applications include some form of automatic compensation for such delays during playback on inserts, so you simply won't be aware of them — but this compensation does not necessarily apply to all signal paths. For instance, Cubase VST from version 3.7 onwards and SX 1.0 do compensate for such latencies in insert paths, but not for aux sends or groups, which may leave untreated audio tracks 'ahead' of the treated ones by an amount equal to the extra DSP latency path.
To combat this limitation, it's possible to delay the untreated audio tracks, to bring them back into sync: simply drag the tracks in question to the left by the appropriate amount. Some applications provide dedicated track-delay parameters, calibrated in milliseconds, to make this process easier, but an alternative method that's probably less prone to user error is to insert a delay 'compensation' plug-in, that simply adds the appropriate delay, to those tracks not using any DSP plug-ins. Universal Audio developed the UAD Delay Compensator plug-in to do just this for UAD1 users.
Further complications may arise when you have both MIDI and audio tracks in the same song. In this case, delayed audio inserts may be automatically compensated for by your music application, but the MIDI tracks may then end up ahead of the audio ones. You can, once again, move these tracks by dragging. (Universal Audio have a track advance plug-in to instead move the audio tracks forward by the appropriate amount.) Fortunately, the latest versions of many VST host applications, including Cubase SX 2.x, Nuendo 2.x, Magix Samplitude 8.x, and all DirectX host applications, including Sonar, perform comprehensive compensation for all signal paths, so using VST or DX-compatible DSP plug-in hardware becomes far more transparent to the user.
However, Emagic's Logic Audio still only compensates for its tracks and channels, but not busses and auxes, and as development on the PC version is now at an end this situation, unfortunately, won't change. If your audio software doesn't provide compensation, you could try downloading the AnalogX SampleSlide (www.analogx.com/files/" target="_blank">http://www.analogx.com/files/ sslide.exe), a tiny 192K DirectX plug-in that lets you delay mono or stereo audio streams by a specific number of samples. As most sequencer applications (including Logic Audio) automatically compensate for insert effects, the most common SampleSlide scenarios are when you want to apply DSP reverb to multiple tracks using effect sends, or DSP/native plug-in compression to several tracks via a group channel (many native dynamics plug-ins look ahead in the waveform to anticipate peaks, and therefore impose a delay, just like DSP-based ones). This will result in the reverb return or compressed sounds playing slightly late.
The solution is to re-route the outputs of all the tracks, except the one playing back the DSP return signal, from the main output channel to a group channel or different output buss. Then use SampleSlide as an insert on this group or buss. If you type the DSP delay, in samples, into SampleSlide, all the tracks will then end up back in sync.
Effects & Monitoring Live
Live monitoring with DSP plug-in (as opposed to on-board soundcard) effects remains a problem area, since the signals are arriving in real time, giving no opportunity for compensation. In this scenario, the only thing you can do is try to run your soundcard at its lowest possible buffer size, thereby producing a low latency value, bearing in mind that any DSP plug-ins you apply to an incoming signal will be subject to at least double this latency before you hear the results.
If your PC is struggling to manage a low enough latency for comfortable monitoring when recording, while at the same time playing back audio tracks using plug-ins, there is a way around this: temporarily mute most of the tracks and deactivate their plug-ins. Then you should be able to reduce your latency value during the recording and raise it again for mixing. Steinberg have also introduced the 'Constrain Delay Compensation' feature in Cubase SX 2.01, which temporarily disables compensation during VSTi and live audio recordings, to minimise delays.
Ultimately, it will always remain next to impossible to automatically compensate for plug-in delays in a multitasking computer environment, while at the same time providing low-latency input monitoring, since the two approaches are mutually exclusive. However, one possible solution would be to try running a dedicated DSP card such as the PowerCore or UAD1 alongside a DSP-assisted soundcard such as the Mixtreme or original Pulsar. This would allow 'zero latency' monitoring with DSP effects on your live input signals, with the option of further high-quality delay-compensated plug-in insert and send effects — possibly the best of both worlds!