SC-1 Mic Preamp NOW ON SALE!!!!

After many months of development, testing, troubleshooting, frustration, problems, and accomplishments... the SC-1 mic preamp kit, PSU-1848 power supply kit and power transformer kits are now available for sale!

Click here to ORDER

Photo of a finished SC-1 mic preamp kit!  (Note: XLR jacks and wires shown on picture not included in the kit.)

Features:
- Low-Noise, transformerless, High-Quality Mic Preamp
- All solid-state design, using chipsets from THAT Corp and Burr-Brown
- Soft-start, slow ramp-on +48V phantom power
- Crystal clear Red LED indicator for phantom power
- Electronically balanced input and output stages
- 12-position Grayhill gain selector switch
- Gain range from +6 to +72dB, in 6dB increments
- Input RFI protection
- Input clamping protection
- Output surge protection circuit
- Output RFI protection
... read more...

Download PDF file of Assembly Guide/Kit Instructions

Related products like the PSU-1848 Power Supply Kit, Power Transformer Kit, Power Control Kit are also available for sale. Click here.

 Saturday, December 23, 2006
Saturday, December 23, 2006 10:37:17 PM (Central Standard Time, UTC-06:00) (  |  |  |  )



Compressors are one of the commonly used plugins when mixing/mastering. It's not hard to understand how they work... if you know what each button/knob does and how it affects the sound.

When compressors are used properly, the effect shouldn't be noticeable. ONLY when you compare the uncompressed and compressed signal should you notice the difference in the dynamics. Compressors are often used during tracking or mixdown. And during the mastering process, the whole stereo mix may be compressed and/or limited. 

The most common  controls in a compressor are the THRESHOLD, RATIO, ATTACK, RELEASE and MAKEUP-GAIN. Everything else is fluff... like input/output VU meters, or soft-knee/hard-knee, limiting on/off. 

Think of a compressor as an automatic variable volume control. The "volume" control's behavior is set by the attack, release, ratio knobs. This automatic volume control reduces the dynamics of any audio material that goes above the threshold. And this is where the paradox is. If compessors are used to reduce the volume/dynamics of music, how can it make the music sound loud? Well, the answer is in the MAKEUP-GAIN, where the average level of the music can be raised higher without clipping (chances for clipping are reduced because we just reduced the dynamics!).

So let's discuss the important parameters/control in a compressor...

1. THRESHOLD - This is measured in dB. It's a negative value because we're setting levels below the 0dB mark. Threshold (in dB) is that level at which the compressor should start reducing the output level. So if you set the threshold to -37dB (like in the picture above), signals below -37dB in levels remain unaffected.  Signals above -37dB (or in other words, louder than -37dB) will be reduced by the compressor. By how much should it be reduced? That's a function of the RATIO setting.

2. RATIO - So if a signal exceeds the THRESHOLD value, the compressor should start reducing it. Reduction is measured in ratio instead of a fixed numeric value. So let's say you set a RATIO of 2, any signal increase above -37dB in our example above will be reduced by half.  So 6dB above -37dB will be reduced to 3dB. A high ratio will reduce the the levels by a higher amount.

3. ATTACK - So we now understand that input levels above the THRESHOLD value will be reduced. This means, the compressor is always watching the input and detecting it's levels. The question now is, if a signal exceeds the THRESHOLD level, how fast or slow should we take action (of reducing it)?  This is a function of the ATTACK setting, measured in milliseconds. A fast attack (i.e. low millisecond) will mean the compressor will take action faster than a slow attack speed. Therefore, a slow attack (higher millisecond) setting means the signal that peak may be allowed to pass through. What does this mean? It means the dynamics of the signal may be allowed to pass through normally without being compressed... think of a snare hit on a drum. We don't want to kill the initial dynamics of the snare hit. 

4. RELEASE - Now, if a signal exceeds the THRESHOLD level, at some point in time it will go down below  the threshold level again. If a signal goes below the THRESHOLD level, at what point should we stop compressing? Should we do it as soon as it goes below the THRESHOLD, or a few milliseconds afterwards? This is a function of the RELEASE setting, measured in milliseconds.  A longer setting, i.e. a slow release means the compressor is still holding a "grip" on the signal, i.e. still reducing it. A short or fast setting means the compressor will let go of the signal. Long settings (i.e. a slow setting) produces a gradual less noticeable effect in the sound. Short release times are good for percussion instruments, and long settings are good for vocals. 

5. GAIN - This is also known as MAKE-UP GAIN. If you set a low THRESHOLD value, and a high RATIO amount, the overall signal will be reduced.  So we would want to amplify that signal and this is the function of the GAIN setting. It boosts the compressed signal to a sufficient level that we want.

Great, now that we know the different settings. But what can we use compressors for?

Compressors can be used to "even" out the volume by reducing the peaks in the signal (thanks to the RATIO control).  For example, your singer has bad mic technique and their volume varies a lot. We can use compressors to even out the levels so the vocal levels don't vary a lot.

Compressors can be used to avoid overloading/peaking during recording. (Of course, when used during recording, the change is permanent.)

By setting the appropriate ATTACK and RELEASE settings, we can add punch to our drums, adding impact to our beats.

And compressors are also used in mastering (sometimes too much) which adds punch and volume to the track.  But if a compressor is overused, you could also end up with a crunched sound... losing all your dynamics which makes for a "boring" listen. So don't overdo it. Sometimes, slight compression settings is all that is needed.

Sample Procedure for Compressing

1. First, set the bypass switch on the compressor. Listen to your levels... note how loud they are. Take a look at your VU meters and note the peak signal level.

2. Then take off the bypass (i.e. let the compressor affect the signal).

3. Set the ratio knobs... 2:1 or 4:1, depending on how much you want to reduce the levels

4. Set the attack and release time.  Let's use a fast attack time of 5ms and slow release time of 100 or 150ms.

5. Now, set the threshold level until the compressor is showing a gain reduction of 4, 6 or 8dB.

6. Now, use the make-up gain and set it so that we're showing the same peak signal as before compression (see step #1).

At this point, the sound should be punchier. It's like magic... they both have the same peak signal levels but the compressed signal has more weight and feels more solid.

If the sound is too compressed, and all the dynamics of the signal seemed lost, raise the threshold level (which means you have to lower the makeup gain), or increase the attack time (i.e. make it slower to react). 

The Bypass button is your best friend. Sometimes, subltle compression is all you need so make sure to hit the bypass on/off button often to hear the difference between no-compression and compression.

If you're using compression for the final mix on a stereo track, you may want to combine the compressed and non compressed signal.  This will make the music "thick" while still retaining the dynamics of the original music.


Typical Compression Settings

These are just typical settings, and don't make this the "rule." The exact setting will depend on the material you're compressing and the effect you want to achieve.

Delicate Vocals - We want to set the RATIO to a high value and high TRESHOLD. By using these settings, the softer vocal sections will remain uncompressed and only loud, ear shattering vocals will be compressed.

Threshold - set the threshold (i.e. lower it) so that the loudest vocal section is reduced by 6dB.
Ratio - Set to 6:1 reduction

Make sure threshold setting is not too extreme. We only want the loud vocal sections to be compressed by 6dB and quiet vocals remain uncompressed.

Pop Vocals/Commercials (yes, those annoying TV and radio commercials) - Okay, we want obvious compression here. So we want compression always going on for almost the entire material, and a very big reduction during loud masterial.

Threshold - set the treshold so the the softest vocal section is reduced a little bit. (-1dB or -2dB)
Ratio - Set to 2:1 reduction
Attack - set to fast setting
Release - set to a little bit slow
Make-up gain - increase gain to increase volume again.

At these settings, we're raising the average level of the whole material... i.e. making everything sound loud!

Drums - or any percussive instrument. We want a punchy, thick sound in the mix.
Threshold - set threshold so all drum materials are compressed by -3dB.
Ratio - Set to 4:1
If you have a hard/soft knee switch, switch to "soft knee"
Attack - set to a fast setting
Release - set to a mid setting

By lowering the threshold setting, you can make the drums more compressed. But be careful not to overcompress it too much or you'll lose the dynamics and punch.

Bass guitar/bassy synths - we want a fairly constant level for the bass material so we have a "thick" sound in the mix.  We don't want bass volume going loud and soft, we want it consistent.
Threshold - set so that only loud bass material (i.e. the peaks) are compressed.
Ratio - Set to 4:1
Attack - set to slow/mid
Release - set to slow/mid

Electric lead/rythm guitar - we want compression on these material. We want the guitars to be punchy and level out the volume.
Threshold - set to a low setting, to achieve constant compression
Ratio - Set to 6:!
Attack/Release - play with it




 Saturday, December 16, 2006
Saturday, December 16, 2006 10:36:08 PM (Central Standard Time, UTC-06:00) (  |  )
Looking to save some money? Create your own cables! By spending a few dollars on jacks, shielded cables, solder and soldering iron, you can save a ton of money in cables.

It's easy. Here are some step by step photos. 

At the bottom of this article, is a wiring guide for XLR to XLR, XLR to TRS, XLR to TS, and whatever combination you can think of.

On this guide, I'm wiring up an XLR cable.  First, we disassemble the cable by unscrewing the lock screw. Pull out the rubber plug (black thingy) and then you'll be able to push out the 3-pin connection assembly.



 Wednesday, December 13, 2006
Thursday, December 14, 2006 3:04:02 AM (Central Standard Time, UTC-06:00) (  |  )
DYNAMIC RANGE

The simplest explanation of Dynamic range is the difference between the loudest and quietest parts of the music, as measured in decibels, is called the Dynamic Range.

If you need a review of what is a decibel, click here.

However, there is a dilemma here. How do you define what is the quietest part of the music? It seems an easy definition, but if you consider that electronic audio equipment have inherent self noise, then the quietest part of the music is fainter than the self noise produced by the equipment. i.e. the quietest part of the music is below the "noise floor" of the equipment.

So in this case, the dynamic range is the difference between the loudest part and the noise floor, in decibels.

Are you still confused? okay.... let's say you're attending a symphony concert. Sometimes, the orchestra will be playing full blast! Let's consider this the "loudest part of the music."  Then in one part of the music, only the flutes can be heard... let's call this the "quietest part of the music."  The difference in volume between the solo flute and the whole orchestra playing is the "dynamic range" of the orchestra.  But by some bad luck, you're seated to an obnoxious person that's talking on his cell phone.  You can't hear the flute solo clearly because your seatmate is making so much "noise."  That obnoxious guy, is your "noise floor."  So that means, the dynamic range of your music has been reduced. Because now, all you can hear is anywhere between the loudest part of the orchestra playing and the noisy guy talking on his cell phone.

The above illustration illustrates how "noise floor" can rob you of available dynamic range. 

In real world scenario, we'd want equipment that gives us the lowest noise floor possible. Because this makes the dynamic range available to us bigger.

 Sunday, December 10, 2006
Monday, December 11, 2006 4:00:04 AM (Central Standard Time, UTC-06:00) (  |  )
Just a continuation of the saga of building my 1176 clone compressor/limiter. Click here for Part 1

Parts arrived a few days ago, and other than the audio input and output transformers and chassis, I think I have enough parts to build (2) units. Here are some pics...

First, we have the output LL5402 Lundahl transformer and the LL1540 input transformer. Made in Sweden.  These things are small.


 Thursday, December 07, 2006
Friday, December 08, 2006 3:51:18 AM (Central Standard Time, UTC-06:00) (  |  |  )
With its exciting new music creation tools, powerful customization and speed-enhancing features, and streamlined post production workflows, Pro Tools® HD 7.3 software is a creativity-charged upgrade that’s built for speed, empowering you to work smarter and faster than ever. Whether you’re composing music or mixing audio for post production, this essential upgrade is packed with a wide range of new features and enhancements to help you make the most of your creativity.

For more information about Pro Tools HD 7.3 software, please visit the Pro Tools HD 7.3 product page.

This upgrade is a full, downloadable installation of Pro Tools HD 7.3 software for qualified computers running Windows XP or Mac OS X 10.4 (PowerPC- and Intel-based Macs). For detailed compatibility information, please visit http://www.digidesign.com/compato. This upgrade does not require a previous installation of Pro Tools software.

If you purchased a Pro Tools|HD or ICON system or Pro Tools HD 7.2 software upgrade on or after November 1, 2006, you may be eligible for a free upgrade to Pro Tools HD 7.3 software. To find out if you qualify, please check the email account you used when registering your product. Email notification will be sent to qualified purchasers with information on how to receive a free upgrade to Pro Tools HD 7.3 software.

Pro Tools HD 7.3 software is not compatible with Pro Tools LE™ systems (such as Digi 002®, Digi 002 Rack™, Mbox® 2, and Mbox 2 Pro) or Pro Tools M-Powered™ systems (hardware peripherals from M-Audio).

There is a known sync issue with simultaneous audio scrubbing in Pro Tools HD 7.3 software and MachineControl. Learn more.

Secrets of the Pros - DVD Pro Tools Vol. 1

And while we're at the subject of ProTools, you may want to get your hands on this as a Christmas gift for yourself  (or your studio). With over 4 hours of information this 2-DVD set will not only show you the basics, but also take you much further into advanced techniques for recording, editing, MIDI, and more. You will see how the software and hardware work, and be shown in detail exactly how to use Pro Tools like a professional.

It also goes step by step through Beat Detective which is a powerful part of Pro Tools that allows you to tighten, fix, and even change the groove of drum and percussion parts.

Your host, Ken Walden, is a former 9 year veteran Digidesign/Pro Tools Product Specialist and Tech, and a seasoned engineer with credits on recordings by several top name artists. Ken delivers all the essentials you need to set up your system and begin recording your music without compromise. Learn the ins and outs of creating and honing Pro Tools sessions, and discover techniques that will give you a huge head start.

Topics
Recording (well refined techniques to make recording easy, and keep the creativity flowing)

Editing (methods used by the best in the industry)

Mixing (signal routing, plug-in info, and more...)

Beat Detective

System set-up (how to get the most power and stability from your system)

MIDI -- featuring Reason (set up, signal flow, and detailed editing info)

Software overview (This will take your through the important parts of the Pro Tools software, and show you how to use thes...

Secrets of the Pros - DVD Pro Tools Vol. 2

And when you're done viewing Volume 1, check out Volume II. This DVD picks up where its highly successful prequel left off. It not only dives deep into advanced mixing and editing techniques, but also includes an overview of exciting new 7.x software features, and provides clear instruction pertaining to file management, latency, buffer settings, and more.

Contents
Advanced Pro Tools DVD Volume II is geared toward those who have some previous Pro Tools experience. Explained with the same clarity as the other Secrets of the Pros DVDs, this volume addresses a variety of the more powerful and esoteric Pro Tools features and recording techniques.

With over two hours of information, this DVD will take you to the next level in terms of your Pro Tools-based audio engineering capabilities and understanding.

Hosted by Secrets of the Pros founder Ken Walden, Advanced Pro Tools DVD Volume II provides a wealth of knowledge and technical prowess from a power user's standpoint. These insights will not only help you to get your head around the whole of Pro Tools, but you'll find yourself moving through projects more effectively and efficiently, making the most of your creativity.

Topics
Advanced Mixing -- More techniques to take you deeper into the Pro Tools mixer

Advanced Editing -- Expert level features that will help release your creativity

Pro Tools 7.x Overview -- An overview of what is new and cool, and how to best use these features...


Thursday, December 07, 2006 3:50:04 PM (Central Standard Time, UTC-06:00) (  |  |  )
Okay, here's the second batch of Cubase video tutorials. Grab some cola/coffee/jolt, sit back and relax.

#7 Cubase Tools


#8 Cubase VST Connections


 Tuesday, December 05, 2006
Wednesday, December 06, 2006 4:05:17 AM (Central Standard Time, UTC-06:00) (  |  )
For all you Cubase fans out there... here are some video tutorials to help you make the most use of the software.

#1 Audio Midi Setup in Cubase

#2 New Project


Tuesday, December 05, 2006 3:16:28 PM (Central Standard Time, UTC-06:00) (  |  |  )
Designed to bring the legendary sound of SSL within your audio workstation, the award-winning Duende audio processor is now available for PC. Click here!

With a simple FireWire connection, Duende delivers up to 32 channels of processing at sample rates up to 96khz. The EQ and Dynamics Channel provides a single plug-in slot EQ & Dynamics processing solution with all the power and processing flexibility of an SSL console channel.

Key features include:

EQ & Dynamics Channel

4-band EQ, two shelving sections and two parametric
Variable low-pass and high-pass filter
Switchable EQ characteristics between E Series and G Series EQ
“Over-Easy” soft ratio compression characteristic for smooth transitions
Variable process order routing
Dynamics side-chain processing with independent side chains for compressor and expander/gate
Bus Compressor

The Duende Bus Compressor delivers the punch and drive of the classic SSL Master Bus Compressor, a key element of many legendary recordings. Widely regarded as ‘audio glue’ for a mix, its sound is an essential component in creating a great mix.

More information and the v1.5 software is available at: www.solid-state-logic.com/duende
 Monday, December 04, 2006
Monday, December 04, 2006 11:01:52 PM (Central Standard Time, UTC-06:00) (  |  )

I've seen the new Zoom H4 Handy Recorder. I think it's a neat little thing! And best of all, it's "cheap" at $299. Of course, "cheap" is a relative word, but compared to other gear out there, this seems to be one of the low priced new entry. It looks good too!

While reading the H4 specs, this caught my eye... "The H4 features 2 studio-quality electret condenser microphones configured in an X/Y pattern for true stereo recording."  The keyword here is "electret microphone." And the picture seems to confirm that it's really just an electret mic.

So I'm thinking, maybe I can DIY my own stereo mic.

I've used an electret microphone on a non-audio project. I made a sound-triggered flash sync for my Canon EOS30D camera. In this case, I just used the audio output from the mic to trigger an SCR, which then triggered the camera flash.

Can we used the same electret mic to create a good enough sounding stereo mic? I think we can.

So I begin adapting my sound-triggered flash sync and while doing some Googling around, I came across this product.
Stereo Super Ear Amplifier Kit

It's a kit made by Velleman and it uses 2 electret mics, and a headphone output jack. Hmmmm.... interesting. So I ordered a kit (hey, it's less than $10 so if it didn't work, it's not too much of a loss). You can use the link above to order your own kit.

The circuit is simple... the Left and Right channel is the same, so the operation of the circuit is identical. Basically, an electret condenser microphone is biased for operation using a single resistor and capacitor. The output of the microphone goes to a potentiometer which acts as the volume control. This is then amplified by the NE5532 opamp, which is then routed to a headphone jack. Since the whole circuit operates from a single supply voltage (4.5Volts), the opamp uses a virtual ground by the use of 2 resistors in it's input pin. This of course, means that each leg of the split power supply will be only Vcc/2, where Vcc=4.5Volts.

I built this kit in less than an hour... while watching TV. And construction isn't hard and it's very simple. So if you're ready to jump into this Electronics hobby and looking for your first audio project, why not try this stereo mic kit? It's only less than $10.

Here are some pics during construction... enjoy.
 Saturday, December 02, 2006
Saturday, December 02, 2006 8:29:00 PM (Central Standard Time, UTC-06:00) ( )
What is the difference between dBu and dBV?

If you've followed my previous articles, dBu is 0.775 Volts at 0dB.

But just to let you know, dBu is the same as dBv (with the lower case "v"). i.e. 0dBv = 0.775 Volts. The National Association of Broadcasters adopted the dBv term for 0.775Volts reference = 0dBv

Now, don't confuse dBv (lower case v) with dBV (upper case V) which is referenced to 1 volt = 0dBV

Now, the good news is today, dBu (with the letter "u") is the preferable usage. That is good... and that would avoid confusion with the use of the letter "V" or "v". 
 Friday, December 01, 2006
Saturday, December 02, 2006 12:26:03 AM (Central Standard Time, UTC-06:00) ( )
It's that time of the year... here are some gift ideas for your home studio.

Furman PL8II Rackmount Power Conditioner with Lights
So you have thousands of dollars in gear and you're plugging it "naked" to your wall outlet or with a $5 Walmart power strip? I recommend you buy this power conditioner that will give you added protection to your gear. All your gear plugs to the back and you have a single switch to turn on all your gear. The lights are very handy too illuminating your rack gear so you can see what you're doing.



Korg MA-30 Metronome
Keeping time is important. And this little device from KORG packs a lot of sophisticated features at a small size. You can do rythms such as triplets and quadruplets with inner beats removed.


 Tuesday, November 28, 2006
Wednesday, November 29, 2006 4:52:07 AM (Central Standard Time, UTC-06:00) ( )
Let's continue our discussion of decibels.

When you look at equipment specs, you often see the term dBm. What does the "m" stand for?

The m refers to 1 milliwatt, which is a 1/1,000th of a Watt.

Now, in audio, it's assumed that there is a 600ohm load when you measure this power. Using ohm's law, 0.775Volts (rounded) applied to a resistance of 600ohms, results to about 1 milliwatt of power.

So if you buy a preamp or mixer, and it says it's maximum output level is +20dBm, we can compute that it's maximum output level is 100 milliwatts.

 Monday, November 27, 2006
Monday, November 27, 2006 8:56:22 PM (Central Standard Time, UTC-06:00) ( )
The term decibel has been used and abused and thrown around. You go to a music store and the clerk will tell you, "Buy this one instead, it has -10dBu inputs! " Meanwhile, you nod your head... thinking, "this must be better than the 4dBu unit I saw at the HomeStudioGuide.com website, and it's even cheaper...."

prrrrrrrrrrrttttt!!!

First, let's understand what the word "decibel" really means.

 Sunday, November 26, 2006
Sunday, November 26, 2006 9:42:33 PM (Central Standard Time, UTC-06:00) ( )
Download FREE orchestra samples from this huge database. The vision for The Sound Exchange came about from research originally carried out for the Philharmonia Orchestra by Olivia Lowson in 2000. She asessed the potential for the Orchestra to create an Education and Access Web Site. The possible new site would need to to provide opportunities for people to learn about the Orchestra, about music and at the same time to exploit the internet to help widen access to the Orchestra, which is something the Orchestra was very keen to do.
 Saturday, November 25, 2006
Saturday, November 25, 2006 9:08:05 PM (Central Standard Time, UTC-06:00) ( )
I haven't DIY'ed for a long time, so to kickstart this hobby again, I decided I'm going to build an 1176 Compressor.

The 1176 is one of the classic compressors in used by almost every recording studio out there. (The other classic compressor is the LA-2A). Bill Putnam designed the 1176, based on the 1108 preamp. Back in 1966, the Field Effect Transistor (FET) was just developed and he's experimenting with the properties of this new device. Basically, think of an FET as a "voltage controlled variable resistor."  The FET has three terminals. Drain, Source, and Gate. The resistance between the Drain and Source can  be varied/controlled by the Gate voltage. So what Mr. Putnam was doing with the FET is using it as a variable resistor to contorl the gain reduction in the circuit.

The output stage of the 1176 is a Class A line level amp, feeding a load of 600 ohms. It's got an output transformer which converts the output from unbalanced to balanced, and also do impedance matching. 

If you don't know or don't want to DIY, you can buy an 1176LN (for Low Noise) compressor for about $1800. Or, you can purchae "virtual" 1176 plugins from $130 to $225 price range. Click here for software plugin versions of the 1176.


 Friday, November 24, 2006
Friday, November 24, 2006 9:43:21 PM (Central Standard Time, UTC-06:00) (  |  )
Wouldn't it be cool if you can setup a MIDI network without all the cables and instead just use your local area network? And woudn't it be twice as cool if it's a wireless network??? Here are the step-by-step procedure for setting up MIDI over LAN using Apple's built-in MIDI over Lan feature. Yes! Apple has built-in MIDI networking and I bet most of you didn't know that.


 Thursday, November 23, 2006
Friday, November 24, 2006 3:10:28 AM (Central Standard Time, UTC-06:00) (  |  |  )
A few years ago, the common wisdom was to "record as hot as you can." By HOT, I don't mean temperature wise. The wisdom at that time was to record your signal (from your guitar, keyboard, mic, etc.) so that the on-screen digital VU meters on your ProTools, Logic, DP, or Sonar is just hitting 0 dB, but not going over. Everybody knows that as soon as the red light turns on, you'll get clipping and digital clipping sounds nasty... like a fingernail scratching on a blackboard. ... The above argument sounds LOGICAL. There was even talk "you must use all of the bits as much as possible, so 0dB must be 1111111111111111 and that's good since we used up all the bits, nothing got wasted."

But something got forgotten along the way by the people who advised you to record close to 0dB in your DAW (Digital Audio Workstation... just a fancy name for a fast computer optimized for audio recording).
Thursday, November 23, 2006 5:37:21 PM (Central Standard Time, UTC-06:00) ( )
Digital Performer 5.11 is now shipping. If you own Version 5.0, 5.01 or 5.1, you can download the free Version 5.11 updater here. This maintenance release includes numerous program enhancements and optimizations. The 5.11 update also includes hundreds of new presets for DP5's six included virtual instrument plug-ins.

If you have Kontakt v1.x (Kontakt.component) in your /Library/Audio/Plug-Ins/Components folder... take it out. Move it somewhere else, out of the Components folder...

Otherwise, DP 5.11 hangs/stops/dies on startup.

Anyways, Kontakt 2.2 is out (now a Universal Binary) so no need to keep v1.x

One thing I noticed is DP 5.11 startup and loading of instruments seems faster to me.
 Wednesday, November 22, 2006
Wednesday, November 22, 2006 3:05:11 PM (Central Standard Time, UTC-06:00) ( )
Kontakt 2.2 Update is now available on the NI website. And it's FREE... well, that depends. At first, I've read reports that the 2.2 Update will cost $29. But according to the email NI sent, if you're a PowerPC or Windows user, the update is free. You only need to pay the $29 or 29 Euro update fee if you'll be running it on an Intel Mac (Universal Binary). So I downloaded K2.2 and it seems they also changed the registration and activation procedure. I had to re-activate my copy of Kontakt using their new "Service Center" program. This is a Flash program that runs on your desktop but communicates with the NI server. You can update your personal information, download updates and get technical support using this program.
Wednesday, November 22, 2006 6:16:46 AM (Central Standard Time, UTC-06:00) (  |  )
Just a work in progress.... using a breadboard. API schematics found on the Internet... just google API 312 schematic. You can see the printout of the schematic on the photo. I'm using a Melcor Discrete opamp instead of the original 2520 opamp (which you can't buy unless you have a broken 2520 to surrender to API). Although there are many substitute opamps you can use like the Jensen 990C, Forsell Technologies 992, John Hardy opamp, etc... or if you're really scraping, just use an NE5534 IC chip.
 Tuesday, November 21, 2006
Wednesday, November 22, 2006 2:03:12 AM (Central Standard Time, UTC-06:00) ( )
One of the biggest problems facing home studio owners is money, mulah, $$$$.... more specifically, the lack of money.

Does that mean you can't enjoy this hobby? No, not at all. On this page, I'll list some freebie software (or reasonably cheap software) you can use in your home studio setup.

Now, I understand some of you may be PC users, and some may be Mac users. Don't worry. I'll pick software you can use for both platforms... or if the software is available as cross platform, I'll let you know too.

Audacity


Audacity is free, open source software for recording and editing sounds. It is available for Mac OS X, Microsoft Windows, GNU/Linux, and other operating systems. You can use Audacity to:
  • Record live audio.
  • Convert tapes and records into digital recordings or CDs.
  • Edit Ogg Vorbis, MP3, and WAV sound files.
  • Cut, copy, splice, and mix sounds together.
  • Change the speed or pitch of a recording.
  • And more! See the complete list of features.
** This software even has features like Noise Removal and Change Pitch. Awesome!  And yes, not only you can edit waveforms, but you can also record using Audacity. Current version is 1.2.6 (for OSX) but 1.3.2 is in beta.

More information can be found here. http://audacity.sourceforge.net/

CPS

This software is also FREE and cross platform, available for both PCs and Macs. But note, it does not run on Intel hyperthreading and Intel Macs.

According to their website...CPS is a realtime interactive programming environment for audio, MIDI and other media & I/O. Patches are build by placing objects and making wires between them. Areas of interest are sound design, algorithmic composition, DSP, electronic/computer music and education. CPS can be used in any realtime situation where sound plays a role.


More information can be found here. http://cps.bonneville.nl/


Computer Music Magazine Freebies (UK Mag)

If you visit your favorite bookstore and head to the magazine rack section, look for Computer Music magazine. This is a nice magazine published in the UK full of useful and informative stuff. (Why do US music magazine companies suck? In my observation, US magazines seems like 80% advertisements and 20% content...)

Anyways, back to CM mag. They include a CD with their magazine and sometimes you'd find freebie software in it. Like what? Like the VST CM-101 and CM-303 synth (analog subtractive synth), CM-505 Drum synth, DS-404 sample player, or a step-sequencer called the EnergyCM. They also have the cool looking SR-202 Drum Sampler.

The software are mostly VST based and for PCs only... sorry Mac users. So check the latest issue of CM, and see what software they included in each issue.









ORCA

This new software synth is from Fxpansion, the makers of BFD drums series and the geniuses behind VST-to-RTAS and VST-to-AU converters. BTW, If you're a Mac user, you need to have their converters as these allow you to use VST plugins and instruments in your MAS/AU/RTAS host sequencer like DP, Logic, etc.

Orca is free, all you need to do is register on their site. http://www.fxpansion.com/index.php?page=12&tab=60

They say ORCA is an experimental monosynth based on modeling of analog synthesizer circuits and components. Whoa!

It's available as cross platform for both Windows and Mac (both PPC and IntelMacs). Check it out. It's FREE.




CRYSTAL

Crystal by Green Oak software has been around long and they just keep on getting better and better.  Current version is 2.4.8 and it available for both Macs and PCs.  Crystal is fully tweakable with lots of parameters, has MIDI learn (very useful if you have a control surface). It has both FM synthesis (like the ones used in Yamaha DX7) and subtractive synthesis. Again, this is a VST so Mac users will need a VST converter like Fxpansion's VST-to-AU or VST-to-MAS converter. The nice thing is it's avaialable as Universal Binary so it will work for both PPC and Intel Macs.

If you go to the Green Oak's website, you can also download sound bank patches for free and they have a tutorials section too.

http://www.greenoak.com/crystal/about.html




SampleTank 2 Free

Yes, it's that SampleTank made by IK Multimedia.  It is a fully working version of their SampleTank 2 software with some limitations (of course, so you'll buy the full version) but it's still useable. You just can't import, and you can only use one instance of the plugin.

Now, just because it's FREE doesn't mean it doesn't need to be authorized. But authorization is easy and you just need to fill up a form asking for your Serial #, Name, Digital ID (created by the software), and a real email address where IK will send your authorization code.

And yes, SampleTank 2 FREE is available for both OSX and Windows! So both camps can stay happy and no jealousy. :)

There are some free patches/samples also available on the IK website that you can load into your SampleTank 2 FREE software.

http://www.sampletank.com/Main.html?prod_ST_XL

LinPlug FreeAlpha

Here's another great, and of course FREE VST synth plugin. Sorry, this one is available only for PC users, LinPlug is now up to version 2.03. Per their website, FreeAlpha features a subtractive design to recreate the warmth and feel typically found in early 1980s analog synths.... and yes, it sounds BIG and PHAT. Check it out. 

FreeAlpha is meant to whet your appetite so you can check out the other great software LinPlug makes like the Rob Papen Albino 3, CronoX3, Sophistry Ambient, and others...


 Monday, November 13, 2006
Monday, November 13, 2006 7:35:03 AM (Central Standard Time, UTC-06:00) ( )
Post #1:

Welcome! There are many sites and blogs out there catering to the home recording enthusiast; this one is written to take out the mystery out of making recordings in your home studio.

There are lots of information to be covered. You see, recording is part technical and part artistic process. I'd leave the artistic and musical process to you. I know you have your own vision on how your music should sound. But hopefully, with this site, I can share some technical information, and practical knowledge that will help YOU make better home recordings.

This site will be organized more as a reference site for the newbie, the amateur, or the veteran musician wanting to make his own home recordings. Nothing will be assumed, and we'll discuss the recording process from the ground up. I know some topics will be boring to some of you, and you'd want to start cranking and recording. But if we're going to build a home studio or learn about home recording, we should start at the basics and foundation.

I'm not going to promise you that you'll be the next hot-shot recording or mastering engineer. That's not the point of this site. But I hope that the knowledge you'd gain from this site, you'd be able to produce your own recordings that will sound great and that  you'll be proud to share with your friends, or give to that A&R person.

I'll try to update the site at least once a week, or even more often. I'll try to write something every week even if it's just a short topic.

This is my first post, issue #1, my intro, welcoming you to HomeStudioGuide.com.  This is still a brand new site, but I hope you'll support this site by telling your friends and buddies about it and visiting us once in a while.

 Saturday, January 01, 2000
Saturday, January 01, 2000 7:03:05 AM (Central Standard Time, UTC-06:00) ( )
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