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How to......

last updated: 28/05/09

• ......use your Wii remote
   
  as an alternative midi controller for Ableton. Read More...


• ......generate hooks
   
  and harmonising melodies in Ableton using Eckel.Read More...


• ......generate musical phrases
   
  within Ableton live for full compositions using pure algorithmic processes.Read More...


• ......use a sidechain
   
  compressor in Ableton live.Read More...


• ......Record Vocals
   
  in the Computer Studio.Read More...


• ......pick the right DAW.
   
  Two Industry Standards featured.Read More...


• ......pick the right DAW.
   
  for Apple Mac specific.Read More...


• ......perform live
   
  and which softwares are recommended.Read More...


• ......more Tutorials
   
  check em out!Read More...

How to......

by ian on april 26, 2009

......generate hooks and harmonising melodies in Ableton using Eckel.In Ableton open 2 midi tracks.Drop Eckel into track 1 and your vsti into track 2.In track 1 make "midi from" "No Input".In track 2 make "midi from" "1-midi". Make sure monitor is "in". Then finally press play.Eckel is a MIDI generating VST based on the Euclidean Algorithm. Inspired by this blog entry and the linked PDF.

How To operate
There are 8 instances of the generating part. Pulse is the actual sounding beats, space is the space between the pulses. Speed is how fast they come, 0 is 4th 1 is 8th's etc. Note is the MIDI note that the algorithm spews out.

Download
Windows
OS X (UB)
Manual 
 
Read a lot more of this in the manual, you really should download the manual.

How to......

by ian on april 18, 2009

......generate musical phrases within Ableton live for full compositions using pure algorithmic processes.

• Windows : download Noatikl V1.5 30 day trial software now! [1,933,495 bytes]
  
  
• Mac OS X : download Noatikl V1.5 30 day trial software now! [7,466,421 bytes]
  
  

Noatikl Generative Music Tool - Overview

Noatikl is a powerful, easy to use generative music tool that helps you come up with new musical ideas - and your own generative music.

It uses techniques developed over the last 17 years through our work on the award-winning Koan generative music system, techniques now used in the all-new and superior "Noatikl".

Ableton Live and the Noatikl VSTi

The steps for using Noatikl VSTi under Ableton Live are as follows; these steps apply to both the Windows and Mac versions of Ableton Live.

Configuring the Noatikl VSTi the first time, for use with Ableton Live

When you've first copied the Noatikl VSTi to your computer, be sure to scan for the VSTi from Ableton. Once done, you should see noatikl_VSTi_win (Windows) or noatikl_VSTi_mac (Mac) in Ableton's list of "Plug-In Devices".

Using the Noatikl VSTI within your Ableton Live Project

Step 1: Add the Noatikl VSTi to your project Add the Noatikl VSTi by selecting the plug-ins view, and dragging noatikl_VSTi_win (Windows) or noatikl_VSTi_mac (Mac) from Ableton's list of "Plug-In Devices" on to a MIDI track. Show the input/output options for your tracks; set the track with Noatikl on it, such that input is set to "No Input". Show the Noatikl editor window, and set-up a file to play (e.g. create a couple of voices). Set the Noatikl MIDI Output to be a spare MIDI port; such as MIDI Yoke NT 1 (Windows) or IAC Port 1 (Mac); you can then hide Noatikl if you want to clear some screen space.

Step 2: Add some Software Synths! Create another MIDI track. Drag a VSTi synth onto that track. Make that track's MIDI Input to be the MIDI Port to which Noatikl is sending its data; such as MIDI Yoke NT 1 (Windows) or IAC Port 1 (Mac) in this example.

Step 3: play and enjoy!

Start it all playing! If you can't hear anything, make sure that the green "on" button was set in the bottom panel in Ableton where Noatikl is shown. If this isn't enabled, you won't see Noatikl flash away when playing notes; so that is a clue to if you need to press this button or not!

How To......

by ian on may 28, 2009

......use a sidechain compressor in Ableton live.
The whole french house scene is practically defined by a signature sound. While the music itself is very different, every track seems to have this really loud sound. Some DJ’s call it a “push, pull”, “breathing”, or “suction” sound. It isn’t just artistry that creates this signature sound, it’s a lot of studio magic. That magic has a name: Sidechaining. After reading this tutorial, you’ll know exactly how to do it.

To follow this lesson you will need:

A VST Host (e.g. Ableton Live, Cubase, Pro Tools, etc.)
SSS Stereo Side Chain Compressor - Free VST Compressor (now for Windows and OS X).
Click here to get the SSS Stereo Side Chain Compressor.Windows Version
and
Click here to get the SSS Stereo Side Chain Compressor.OSX Version

What Does Sidechaining Do?

Sidechaining takes one sound and uses it to manipulate another sound. Most often you will see it on compressors, gates, limiters, and expanders. However, it can also be found on vocoders, synthesizers, and other effects. It is a really, really useful feature on compressors and gates because it allows you to place multiple instruments in the same frequency range without clashing. Or, more succinctly, it can make your mix sound really good.

This tutorial is going to demonstrate ducking using a sidechaining compressor. Ducking is the technique used by french house producers to get their characteristic “pumping” sound. Here are two snippets from popular french house songs that make heavy use of ducking:

Daft Punk - One More Time
Thomas Bangalter & DJ Falcon - Together

Listen to the horns in the Daft Punk sample. Notice how they become quiet when the kick drum plays? That’s ducking. The horn volume is ducked to make room for the kick drum. Listen to the DJ Falcon sample. The vocal loop gets ducked for the kick drum in the exact same way.

Now go grab any french house record from the last eight years. Almost all of them do it.

How Does Sidechaining Work?

Sidechaining really just means feeding sound into an effect that works on some other sound. But when most people talk about sidechaining, they are really talking about using sidechain compressors and gates to produce ducking. That’s what you will do today. This tutorial will guide you through setting up a Side Chain Compressor.

So, what does a Side Chain Compressor actually do anyway? It uses one of your audio tracks to control the volume of another audio track. To get a ducking sound, it will lower the volume of one of your audio tracks when the other gets loud. The screenshot below shows the volume of a synth pad before it is effected and after it is ducked to make room for a kick drum.

Now that you know how it works, it’s time to do it yourself. Read on.

Setting Up the Audio Tracks

First you need to prepare two audio tracks to feed into the compressor. The first audio track will hold a four-on-the-floor kick drum loop. The second audio track will hold a synth pad loop. Click the link below to download a ZIP file containing the two loops.

Click here to download a ZIP file containing two WAV audio files.

Start your VST Host (e.g. Ableton Live) and create a new song. Then open the ZIP file and place the two loops on separate audio tracks in your VST host. Make the loops repeat at least a couple times. Label the audio tracks if you prefer. When you are done, your VST host should look like the screenshot below.

When you press play you should hear the kick drum and synth pad playing on top of each other. The goal is to get the pad to “duck out” for the kick drum. You’ll need the SSS Side Chain Compressor to make that happen. Read on to learn how to set it up.

Setting Up the Compressor

Before you can use the compressor, it must be installed in your VST plugins folder. If you haven’t already done that, do that now. Next, load the SSS Side Chain Compressor effect onto the track with the synth pad. If you are using Ableton Live, you can click here to read a tutorial on using VST plugins with Ableton Live.

When press play you hear that nothing has changed. Before the compressor can work its magic you need to route the audio from the kick drum track into the compressor. If you are not using Ableton Live, then consult the manual on your VST Host’s audio routing procedures.However,If you are using Ableton Live, then reference the animated screenshot below.

Once you have routed the kick drum audio into the compressor, you are ready to start ducking the pad. The next section explains how to tweak the compressor’s settings to get the french house “pumping” sound.

Tweaking the Compression Settings

(NOTE: DualComp users should click here to read a tutorial on tweaking the compressor.)

Now when you press play you hear the pad ducking out for the kick drum! But, you don’t hear the kick anymore at all. This is easily fixed. Turn the knob on the far right of the compressor labeled KeyVolume. Turn it all the way up to 0.0db and the kick drum will return in full force. Wasn’t that easy?

The rest of the adjusting is up to personal preference. The threshold is a bit low so you may want to move it up to around -13 or -10db. Since the goal of this exercise is just to duck out the pads, you should turn the ratio knob all the way to infinity to give the pads the largest dynamic range possible. Keep the attack low at around 5ms. The hold knob will keep the pads quieter a bit longer and for this exercise should be somewhere around 40ms. The release determines how long the pads will fade in after each attack (and hold) and in this case should be around 90ms. Lastly, there is no need for any makeup volume so the gain knob should stay at 0db. (See the screenshot below.)

If you are using Ableton Live, you can click here to download the finished arrangement file.

Now you are ready to write some funky french house. Or, at least you’re on your way

Conclusion

I hope you enjoyed learning about Side Chain Compression. This technique alone can really improve the quality of your music. Keep experimenting and remember that this tutorial only scratches the surface of what is possible with sidechaining. If you have any questions or comments, please post a message on the MashedUK Forums. Click here to visit the MashedUK forum.

For more advanced side chaining, read this tutorial on side chain compressing a submix in Ableton Live.

How To......

by ian on may 28, 2009

......Record Vocals In The Computer Studio.
First Things First
A vocal recording starts at the microphone, but before even getting into the issues of mic choice and mic placement, there's the matter of the recording location to sort out � and it goes without saying that this should be isolated as much as is possible from the physical noise generated by the computer's fans and drives. A lot of people think they need to buy better gear to sort out a vocal issue, but when you get to the bottom of the problem, it's often down to the room and its influence on the sound.
You don't need to do anything too fancy to record vocals, but the mic should be well away from any walls, and the area directly behind the singer should be non-reflective. This could be an area of foam tiles or it could be a duvet, but one point to watch out for is that, in rooms where a lot of damping material has been applied, you'll often find that it only absorbs effectively down to around 250-300Hz. So what actually happens is that frequencies below 300Hz are allowed to predominate, making the sound seem congested or boxy.

This is a particularly common fault in small vocal booths, where introducing some reflective scattering surfaces helps balance out the low-frequency resonances. Worse still are rooms that have the walls carpeted, as carpet only absorbs higher frequencies, so you end up with a sound that's both boxy and dull. It's often best then to record in an ordinary domestic room, probably with a carpet or rug on the floor and a duvet hanging loosely behind the singer. If the room is still too lively, hanging further absorbers to the sides of the singer will help, but it's very rare that you need to go much further than that.
Set up the mic a couple of feet from the centre of the room and make a test recording using no processing at all to see if the basic tonality is OK. If it's not, the chances are that the problem is with the room or the mic placement, so try more hanging absorbers and move the mic around relative to the walls.

Choosing Your Weapons
Now the choice of mic. I've had people call asking if they should swap their �1000 tube mic for a �2000 tube mic, but in the majority of cases their dissatisfaction with their sound has not been the fault of the microphone. Yes, a good mic does sound better than a cheap one, but a well-designed budget microphone can produce superb results when used correctly � even a stage dynamic mic such as an SM58 can sound great! Although you can get away with using a dynamic mic, the low cost of imported capacitor mics is now such that anyone can afford one, and, as they're a lot more sensitive than dynamic mics, preamp noise is less of a concern, because you don't need to use so much gain to get a recordable signal.

Capacitor mics will give you a more open-sounding top end, and in the majority of cases they're the best choice. Note that some back-electret models � one example being the popular AKG C1000S � are designed primarily for live use, so their sensitivity is more akin to that of a dynamic mic than a true capacitor model. They're fine for recording close vocals, but perhaps less well suited to quiet or distant sound sources. From a noise and tonality point of view, it's probably fair to say that choosing one of the many large-diaphragm cardioid models on the market offers the best balance between cost and quality. Once you've perfected your vocal recording technique, then is the time to start checking out better mics and preamps. It's always best to use a shockmount with any vocal mic, and unless low-frequency noise is a real problem, record without the low-cut filter, as you can do your filtering in software if need be. Equally important is the use of a pop shield, without which plosive 'B' and 'P' sounds are likely to cause popping. These particular vocal utterances are accompanied by a directional gust of air which, in the absence of a pop shield, slams into the diaphragm of the mic like a hurricane into a garden shed and results in a huge low-frequency signal that's often so high in level that it clips the preamp. Trying to shift popping at the mixing stage using filtering is a bit like trying to level a mountain using a garden strimmer � it's unlikely to work! The pop shield should be around three inches from the mic and the singer between six and 12 inches from the pop shield, though better-trained singers may vary their distance to help maintain a more even level.

If there are tonal elements in the voice being recorded that you don't like, try moving the mic relative to the singer before thinking about EQ. Often moving the mic up or down by a few inches helps, and in some cases moving the singer further away works too. On the other hand, if you have a singer with a quiet voice and you need to add a bit more vocal weight, working closer to a cardioid mic can help, as the proximity effect adds some useful bass-end reinforcement.

Making Test Recordings
Once you're at this stage, check the peak levels on your preamp to ensure you're not approaching clipping on any of the loud notes and also check the recording level on your computer or recorder's meters. If you have any tube drive setting or other similar processing novelties, try bypassing them to start with. Now it's time for a trial recording using enclosed headphones to feed the singer their cue mix � open phones spill quite badly into the mic. If you can flip the phase of the headphone mix, try both settings to see if the singer can hear a difference. They often can because of the way the headphone sound combines with the direct sound they hear via bone conduction. Usually the singer will find it easier to sing with one phase setting than the other. Most singers also like a little 'comfort' reverb in the phones, but discuss this with them and set up your system so this is not recorded.

After the test recording has been made, listen carefully to the soloed vocal sound to ensure that it's clear, clean and free from unwanted room resonances. Any natural peaks in the voice that sound too 'forward' can be tamed later using narrow bands of EQ cut, but for now aim to capture an honest representation of the voice and don't be afraid to directly compare the acoustic sound of the voice with your recording so that you can see where any differences occur. You should be able to get to this stage using just about any decent mic and preamp, though a good-quality mic preamp will help you retain the quality you're getting from the mic.



How Much Should You Compress?
Look at the waveform levels in your sequencer and see how much variation there is. If there's a big level fluctuation, you may need to add some compression when recording, but never add more than you'll ultimately need, as you can't take it off once it's been added. I usually stick to ratios of up to 4:1 and then adjust the threshold to give a gain-reduction reading of no more than 6dB on the peaks. Use a fast attack and a release time of around a quarter of a second, or the automatic mode if your preamp's compressor has one. If you don't have a compressor in your preamp, then record with no processing and use a software compressor when mixing. Excessive or inappropriate compression at this stage can lead to a congested, lifeless sound that's almost impossible to fix later. It also pays to bear in mind that compression brings up the effects of the room ambience in quieter passages, so while you may not hear the room on an unprocessed recording, it may start to intrude once you start to add compression. This is why it's so important to fix the recording space before you start recording.

It may be that some sections of the song, often choruses, are sung louder than the verses, in which case you might opt to record them separately on two different tracks so that you can match up their levels more easily. Other options include recording everything onto one track, then dividing that track into its verse and chorus sections before moving the chorus sections to a new track. Doing this makes the levels easier to manage without having to use automation, and you also have the ability to use two different compressor settings (or even two completely different compressors) on the two tracks. Often the louder section will need a higher ratio to keep it even, especially if the section includes a lot of natural dynamics.

Although compressors can compensate for many level problems, I find that there are occasionally words or phrases that are much too loud or too quiet and that compression doesn't completely cure. In this case it's invariably best to use your sequencer's level automation capabilities to balance the levels.

Sitting Comfortably
Getting a well-recorded vocal part to sit in a track relies on it having an even level, the correct tonal and level balance, and the right reverberant ambience. I like to start out by getting the compressor and gain automation settings right, so that the vocal sounds even enough to sit in the track, with none of the words or phrases sounding too loud or too quiet. Where you really want to add density to a voice, try upping the compression ratio to around 8:1 and then adjusting the threshold so that the gain-reduction meter just registers a decibel or two on average-level parts. This means quieter sections will be left alone and anything louder will be jumped upon fairly hard to bring it closer to the average level. The gain reduction that shows on peaks will depend on how much level fluctuation there is in the track � values of around 10-15dB are not uncommon on peak-level phrases. If your compressor has a choice of Peak or RMS limiting, RMS usually works best for vocals.

Next comes EQ, but in most cases you'll need very little unless you are after a special effect, or if the voice has some defects that you need to fix. The general rule is that, while EQ cuts can be made over quite narrow bands, boosting should be as gentle and wide as possible. If you think the voice does have an over-emphasised component, then set one of your EQ's peaking filters to full gain, with a Q value of around 1.5, and then sweep it through the frequency range until you hear it pick out the frequency that's been giving you trouble. You can then apply cut at this frequency, ideally using as narrow a filter (high Q) as possible to get the job done. If more than one frequency band stands out, then engage another filter section and do the same, but always apply as little cut as you can get away with, as less EQ is always better. For more gentle shaping, try gentle cutting or boosting in the 100-300Hz band to add warmth or reduce chestiness. Also check the 1kHz region, as this is where nasal vocal sounds hang out, and if you need to add presence then try a gentle boost between 4kHz and 6kHz. The magical sense of air and space requires a very wide boost centred between 14kHz and 16kHz, but check what you're doing by comparing your EQ settings with the EQ bypass position, just to make sure you haven't gone too far. If you use a pitch-correction product such as Antares Auto-Tune, the most natural results are usually obtained at the slower pitch-correction settings. The default correction rate seems a little too fast to me, as I can hear it working. If you can split up the vocal track into two tracks where the phrases and words that need tuning are all on the same track, then you can use pitch correction only on those sections that need it, which should result in a more natural-sounding end product. Pitch-correction may be applied a little more heavily to individual backing vocals, where any artefacts are more likely to be disguised by the layering of two or more voices.

Reverb Settings & Mixing
That leaves reverb, and in my experience this is trickier to get right than you might imagine. For a start, few plug-in, host-powered reverbs have the clarity and character of a decent hardware or DSP-card reverb. However, if a plug-in is all you have, pick the best algorithm on offer and set up the reverb in your virtual mixer's aux send/return loop (with the effects mix set to 100 percent wet) so that you don't have to duplicate it for each vocal part. This will minimise CPU drain. If you have a good hardware reverb, you can patch this into any spare soundcard inputs and outputs you have and then route the virtual mixer's aux sends though it � a trick I often do with my Lexicon MPX550. The output of the reverb feeds back into the soundcard and then into a spare stereo mixer input.

Traditionally, vocals are treated using plate or room reverbs, but on the cheaper hardware boxes and most plug-ins, the presets tend to muddy the sound before they produce the desired thickening effect. One strategy that I have found to work well is to use a reverb algorithm where the early reflections level can be adjusted independently of the reverb tail, then turn the level of the reverb tail down by around 6dB. The early reflections have the effect of thickening the vocal without making is sound washy, and by turning down the level of the reverb tail, you can still get away with a fairly long reverb time (typically around 1.8 seconds) without losing clarity. Up to 100ms of pre-delay also helps add depth without clouding the picture, and you may even be able to drop the reverb tail level further if you're aiming for a more subjectively dry sound. If you have a commercial recording in a similar style, keep this on hand as a reference when you're setting up the mix, as it helps to compare general tonality and reverb settings.

The final test is to set up what you think is a good mix, then listen to it from outside the studio door. The vocals should be loud and proud, but not so loud that they sound 'pasted on', and you normally only need enough reverb to make the vocals sound as though they were recorded in the same acoustic space as the rest of the backing track. I've mentioned this 'standing outside the door' trick before, because it really is the best way to judge balance. A problem that might be completely unnoticeable when you're sitting in front of the monitors will jump out at you once you leave the room. At this stage you can make a test mix, burn a CD and then play it in as many different systems as possible, as this might reveal something that your monitoring environment didn't. Sorted � hopefully!

How To......

by ian on may 28, 2009

......pick the right DAW.
Two Industry Standards featured.

Digidesign Pro Tools

Pro Tools|HD � Pro Tools LE � Pro Tools M-Powered

From the largest recording facilities to the smallest bedroom studios, Pro Tools� is the most popular sound creation and production system in the world. Featuring professional hardware and award-winning software that work together with your computer, Pro Tools lets you easily accomplish everything from composing music and recording to editing and mixing sound for broadcast and post production � all within a single environment.

Pro Tools|HD
For the ultimate in audio quality and workflow efficiency, more music and post-production professionals use Pro Tools|HD� systems than any other DAW. With superior sound quality, unmatched processing power, Pro Tools HD software, and the flexibility to scale the system to your needs, Pro Tools|HD provides the power to meet any challenge.

Learn more about Pro Tools|HD systems

• Professional, customizable systems deliver pristine audio clarity, powerful performance (with zero latency), and rock-solid reliability
  
• Work with up to up to 192 simultaneous audio tracks (256 virtual tracks)
  
• Compose and mix music with up to 128 simultaneous Instrument tracks
  
• Fully integrated MIDI sequencing with up to 256 simultaneous MIDI tracks
  
• Supports up to 192 kHz resolution and up to 160 channels of I/O
  
• Gain full hands-on control with an ICON integrated console, C|24�, or Command|8� control surface
  
• Supports Mac OS X, Windows Vista (32-bit Business or Ultimate), and Windows XP-based computers

Pro Tools LE
Get many of the same audio tools the pros use in a more affordable package. Pro Tools LE� systems come in a wide range of pro-quality hardware interfaces, and include Pro Tools LE software, enabling you to create, edit, and mix professional-quality productions right in your home, project studio, or on the road.

Learn more about Pro Tools LE systems

• Flexible audio interfaces delivering amazing sound quality, low latency, and rock-solid reliability
  
• Work with up to 48 simultaneous stereo audio tracks (expandable up to 128 tracks*)
  
• Compose and mix music with up to 32 simultaneous Instrument tracks
  
• Fully integrated MIDI sequencing with up to 256 simultaneous MIDI tracks
  
• Supports up to 96 kHz (audio interface dependent)
  
• Gain hands-on control with a C|24 or Command|8 control surface
  
• Supports Mac OS X, Windows Vista (32-bit Business or Ultimate), and Windows XP-based computers

Pro Tools M-Powered
Bring the power of Pro Tools to your M-Audio� interface or peripheral. Pro Tools M-Powered� is a standalone version of Pro Tools software that works exclusively with supported M-Audio hardware peripherals, enabling you to create, edit, and mix professional-quality music.

Learn more about Pro Tools M-Powered software

• Works with many M-Audio USB and FireWire interfaces, guitar and DJ interfaces, audio cards, and keyboard controllers
  
• Work with up to 48 simultaneous stereo audio tracks (expandable to 64 tracks**)
  
• Compose and mix music with up to 32 simultaneous Instrument tracks
  
• Fully integrated MIDI sequencing with up to 256 simultaneous MIDI tracks
  
• Supports up to 96 kHz resolution with certain M-Audio interfaces
  
• Gain hands-on control with a Command|8 or M-Audio control surface
  
• Supports Mac OS X, Windows Vista (32-bit Business or Ultimate), and Windows XP-based computers

* Achieve up to 128 mono (or 64 stereo) simultaneous audio tracks and 7.1 surround with the optional Complete Production Toolkit. Achieve up to 64 stereo or mono simultaneous audio tracks with the Music Production Toolkit 2 or DV Toolkit 2 option.

** Achieve up to 64 stereo or mono simultaneous audio tracks with the Music Production Toolkit 2 option.

How To......

by ian on may 28, 2009

......pick the right DAW.
Two Industry Standards featured.

Steinberg Nuendo

New Features in Nuendo 4.2

Advanced Integration of Steinberg�s new hardware

Experience a new level of true integration with Nuendo 4.2 and the new Steinberg audio hardware. The MR816 CSX and the MR816 X are a new generation of solutions for audio professionals. Together with the CC121 Advanced Integration Controller, MR 816 CSX and MR 816 X are the first products, melding with Nuendo to form a new level of all-round solution.

Developed by Steinberg and Yamaha and precision-manufactured by Yamaha to the most exacting standards, the MR816 CSX and the  MR816 X are a new generation of advanced integrated interfaces of solutions for audio professionals.
All aspects of I/O setup and configuration, handling and routing are mirrored between Nuendo and MR816 CSX and MR816 X.

• Outstanding Sound Quality
  16-channel I/O FireWire Interface with 8 discrete Class A �D-pre� mic preams
  
  
• True Integrated Monitoring
  
  Thanks to the new True Integrated Monitoring� it�s now easy to create real-time, unique � and most importantly, no-latency � mixes with DSP FX from different audio input sources all from within Nuendo
  
  
• Complete Nuendo user interface integration throughout
  
  MR816 CSX and MR816 X both feature direct integration into Nuendo 4.2 through unique windows and views within Nuendo that give full access to all hardware functionality.
  
  
• Control Room � with no latency
  The Advanced Integration DSP Studio series enable the Nuendo mixing engine to create no-latency monitor mixes, using onboard VST3 DSP FX to include EQ, Compression and Reverb to quickly tailor a mix for a performer.
  
  
• DSP Power for Nuendo
  
  Nuendo 4.2 can take full advantage of the MR816 CSX and MR 816 X onboard DSP chip. Both units feature Yamaha�s renowned REV-X reverb and the MR816 CSX can run up to 8 instances of the first-class Sweet Spot Morphing Channel Strip plug-in, featuring the latest compression and EQ technology. The full DSP capability is also accessible VST plug-in for mixing playback tracks in Nuendo!
  
  
• Quick Connect and Integration
  The unique Quick Connect feature allows for instant routing and rerouting of inputs to Nuendo tracks on the fly from the front panel of the I/O with the push of a single button.

Find out more about MR816 CSX and MR816 X.

 

Steinberg CC121 Advanced Integration Controller

Nuendo 4.2 supports all functions of Steinberg�s new Advanced Integration Controller � the next-generation dedicated USB Controller. CC121 provides totally integrated, full tactile control of Steinberg�s Nuendo.

 

• Advanced Integration Controller Knob - �point&control� for any Nuendo, FX or VSTi parameter
• Dedicated controls for EQ, transport and Channel Settings
• 100mm touch-sensitive motorized fader
• Instant Plug & Play: no additional setup required

Find out more about CC121 Advanced Integration Controller

Improved support of Yamaha Hardware

Yamaha MOTIF-RACK XS / MOTIF XS6/7/8
Completely new and extended support for the MOTIF XS has been added to Nuendo 4.2.2. Nuendo automatically detects any Yamaha MOTIF XS unit which is connected via FireWire (IEEE1394) to the computer.

The detected MOTIF is automatically configured and immediately ready to use as an external VST3 instrument.

A new section has been added to the Track Inspector: Track Quick Controls. This set of eight user-definable real-time controllers can be assigned to any parameter available on the current track. Track Quick Controls are available for Audio, MIDI and Instrument Tracks. These parameters include any mixer parameter for audio and instrument tracks, insert plug-in parameters, VSTi parameters and MIDI controllers for MIDI and instrument tracks. Track Quick Controls are assigned individually for track and saved with the project, as well as in the new track presets. MIDI Learn functionality makes setting up Track Quick Controls with a remote controller extremely easy. Once assigned, remote control knobs or fader functions will follow the track selection.

Track Presets

A further convenient and powerful new feature in Nuendo 4 is the ability to work with Track Presets - especially when they�re combined with the new MediaBay. Track Presets are snapshots of entire track/channel strips. They are available for Audio, MIDI and Instrument Tracks and can be categorized and managed using the MediaBay search engine and attributes such as Category, Style, Character etc. Audio Track Presets contain EQ and insert effect information, while MIDI Track Presets save the MIDI port, channel and patch data, as well as additional MIDI track parameters like MIDI plug-ins. Instrument Track Presets include a combination of all of the above, as well as selected instrument/preset information.
Track Presets open a whole new dimension in sound and track management. Besides being used as templates for new tracks, Track Presets can also be used to extract EQ or FX settings and paste these into other tracks. Track Presets are also available in the Nuendo MediaBay. Multiple Tracks can even be saved into a single (Multi-)Track-Template, allowing swift access to complex track setups for multiple tracks.

New state-of-the-art Channel EQ

The new Channel EQ in Nuendo 4 is one of the key new plug-ins in Nuendo 4. One of the best-sounding �onboard� EQs available today, the Nuendo 4 Channel EQ provides unparalleled sonic detail and flexibility when compared to �standard� channel EQs in other DAWs. The new Channel EQ now provides up to 7 different characteristics per band and features an inverse switch for quick flipping between amplification and gain reduction. The Q factors of the new shelving modes now act as adjustable slopes of the shelf, allowing creation of resonant peaks for more punch and advanced frequency differentiation. This new EQ - also available in the new StudioEQ plug-in - offers great tools to add character and depth to any sound and represent a major advancement in sound quality and usability for Nuendo.

Ready for Intel-based Macintosh computers, Windows Vista 64

Not only does Steinberg continue to deliver a cross platform product in a single Nuendo package but Nuendo 4 is also ready to run on the latest operating systems by both Microsoft and Apple. While Windows XP and the Apple PPC platform continue to be supported, Nuendo 4 also runs on latest Intel-based Macintosh computers and on Windows Vista. The latter, of course, not only in 32 Bit mode, but also as a real 64 Bit compilation allowing the user to scale projects to an unknown maximum and address up to 128 GB of RAM in a single computer. If plug-ins need to be used which are not available for these new platforms, the included VST Plug-In Bridge for Intel-based Macintosh and Vista 64 will start them in a separate process, thus allowing these plug-ins to run side by side with the native ones.

Redesigned Sample Editor

The Nuendo 4 sample editor now features a new inspector-style section to the left. Features that were spread across many different parts of the program are now in one place. This includes tools for audio tempo definition, quantization, audio warp and Hitpoint-related functions.

Project Logical Editor

A new powerful logical editor opens up countless new possibilities for task automation and project-wide �power-editing�. The Project Logical Editor allows manipulation of tracks and parts/events, including folder tracks with actions such as deleting everything beyond the cursor, un-muting all muted parts, renaming tracks/events or even deleting automation of a specific type. And this can selectively be done to any, all or multiple selected parts/tracks in one single step. But the Project Logical Editor comes with a large selection of presets, which are available directly from the Edit menu. In addition, users can create their own presets, which are automatically added to the preset menu. All Project Logical Editor presets are also available as targets for user definable key commands, making it an extremely powerful tool that can significantly speed up production and editing.

Instrument Tracks

Instrument Tracks combine MIDI Input and Audio Output into one single track and mixer channel, and are thus a huge time-saver for using virtual instruments in Nuendo 4. Setting up a track for a VSTi is now only two clicks away. Instrument Tracks are fully integrated with Nuendo�s new SoundFrame Universal Sound Manager, use only a single channel in the VTS mixer and have the additional benefit of showing MIDI and plug-in automation in the same location.

MPEX 3 � High-quality audio time-stretching and pitch-shifting

MPEX3 is the latest and highest-quality MPEX time-stretching and pitch-shifting algorithm available. It offers improved overall sound quality, support for more sampling rates, better performance, and seven different modes to choose from. MPEX3 sounds fantastic and the new modes are optimized for specific audio material, which improves sound quality even further.

New Nuendo EuCon Adapter (Optional)

The optionally available Nuendo EuCon adapter, which provides integration with Euphonix hardware controllers and mixers, has been updated to map the new functions of Nuendo 4 directly onto the hardware surfaces of MC, System 5 MC, System 5 Hybrid and S5 Fusion. The all-new automation system of Nuendo is fully controllable from the hardware. The new routing capabilities of Nuendo 4, including side chaining of the plug-ins and the new recording from busses option, can also be easily controlled. And lastly, the new version of the EuCon adapter further enhances the MIDI implementation by providing access to MIDI routing and the new Instrument channels.

Additional New Features

Nuendo 4 also includes a range of additional new features that can make all the difference:

• VST Plug-in Bridge
• Updated WK-Audio ID Support
• Use Mouse Wheel for Event Volume and Fades
• Fade Handles always-on-top option
• Advanced QuickTime 7 Support
• Separate Goto Timecode command window
• Apple Remote support
• Direct X ASIO 64 Bit
• MP3 Surround
• New Trim Tool
• Load Projects inactive
• Arranger Track: new performance funtions
• Global Track Transpose
• New File Type: MIDI Loop

Detailed information: Additional New Features in Nuendo 4 (PDF, 139 KB)

Important Product Improvements

In direct consultation with users and industry partners, Steinberg has implemented a range of enhancements to existing Nuendo functionality to offer more streamlined and powerful tools in a range of tasks and processes.

• Improvement for Output Configuration of VSTi�s
• Improved Channel Strip
• Improved Waveform Resolution
• Improved Audio Warp Real-time Transpose
• Track Control Context Functions Improvements
• Improvements on the Track Inspector
• Improved Audio & MIDI Hardware Setup
• Track Sheet Printing Improvements
• Time Displays Windows Improvements
• Improved Remote Controller Support
• Improved XML scripts for EuCon mapping of VST plug-in parameters
• ADM (ASIO Direct Monitoring) Improvements

Detailed information: Nuendo 4 Product Improvements (PDF, 130 KB)

How To......

by ian on may 28, 2009

......pick the right DAW.
for Apple Mac specific.

Logic Pro 8

How To......

by ian on may 28, 2009

......perform Live.
and which softwares are recommended.

Ableton Live 8

Ableton Live is about making music. When you're writing and composing, Live is quick and responsive. When you want to record and develop your ideas, Live has the depth and toolset for intricate production. It's got the features to take your DJ skills to the next level and it's stable and flexible on stage, whether you're playing in front of 10 or 10,000 people. However and wherever you make music, whatever music you make, Live will inspire you.