Showing posts with label tech. Show all posts
Showing posts with label tech. Show all posts

05 August 2010

Apollo Studio Tech (Part 8)

In the previous posts I focused mainly on creating and processing of the audio on my PC, but off course it is also quite nice to hear the output somewhere. In my studio I use different sets of monitor speakers to judge my mixes on. In the picture you can see from left to right: Genelec 8250, Mackie HR824 MKI, and Avantone Active Mixcubes. The Genelecs are my main monitors, so they are closest to me and under my desk is also the matching 7260A Sub-woofer. Next to these 3 stereo sets there is also a Surround setup with 5 Behringer B2031A's and a B2092 Sub-woofer. You can read all about why I use these monitors in the equipment section of the studio menu. This article is about the routing of the audio to these monitors.

There are several audio sources in my studio and they all need to be routed to the monitors. On the right you can see the schematic again in a simplified form with only everything on it that is relevant for the monitoring. Two of the audio sources are the Mastering and Video PC and Audio PC. The mastering PC has a dedicated RME Fireface 800 that outputs an SPDIF signal. The Audio PC only has the RME MADI interfaces in there so in one channel it outputs its audio signal to the ADI-648 that sends it to an ADAT channel and the Friend-Chip converts this to an SPDIF signal that goes into an Mutec MC4. This is an Digital Audio Converter with 3 inputs that you can select from and it also has three inputs. You can see three other audio sources on the input from the MC4 that are mixed together in a Mutec Merger. These sources are a Alesis Masterlink HD/CD recorder, an Eminent Multimedia Player that I use to watch movies in my studio and a Logitec Sqeezebox that I use to listen to music in the studio.

The Alesis Masterlink also has an output routed back into it so that I can record everything that comes out of my monitors in the studio without using a PC. Then there is two more outputs from the MC4. One is directly going into the Genelec 7270A Sub-woofer that in its turn sends it to the Genelec 8250's and the other signal is going into a Presonus Central station that I also use as DA converter for the analog stereo monitors. An analog signal is also sent from the Central Station to the SPL Surround Controller so I can also listen to a stereo mix on the Behringer surround set. There is also an analog surround signal coming directly from the analog outputs of the RME Fireface 800 attached to the Mastering and Video PC.

In the picture on the right you can see the Presonus Central Station on the bottom. It is really the heart of the analog audio monitoring setup. It has a volume controller on the right that is right in front of me and next to that you can select monitors A, B and C. On here are the Mackies HR824 MKI and the Avantones Active Mixcubes. I still need to also connect the analog inputs of the Genelecs to this. There is also a main out on the back that in its turn is connected to one of the stereo inputs of the SPL Surround Controller. This output does not respond to the volume controller, so I can set the level for the surround set on the SPL.

In the picture on the right you can see the Presonus Central Station on the bottom. It is really the heart of the analog audio monitoring setup. It has a volume controller on the right that is right in front of me and next to that you can select monitors A, B and C. On here are the Mackies HR824 MKI and the Avantones Active Mixcubes. I still need to also connect the analog inputs of the Genelecs to this. There is also a main out on the back that in its turn is connected to one of the stereo inputs of the SPL Surround Controller. This output does not respond to the volume controller, so I can set the level for the surround set on the SPL.

This is the last part in this series for now. I hope you enjoyed it.

04 August 2010

Apollo Studio Tech (Part 7)

In the past I used only software plugins for sounds effects like reverbs, filters , chorus, flanger and delays, but while mastering my first album at Groove Unlimited I heard how much better their hardware reverbs sounded. So I started looking for some outboard gear myself as well. The advantage of outboard gear is off course that it is dedicated for that purpose only and it saves a lot of CPU usage on the audio PC. Disadvantage is that syncing is a bit more difficult and you have to compensate for latency. But Sonar takes care of that for me. In the pictures you can see my favorite outboard reverbs just below the RME ADI-648 and Apogee Big Ben. From the top down they are: Quantec Yardstick 2402/f, Eventide Eclipse, Bricasti M7 and Lexicon PCM96 Surround. I use at least two of these in almost every track I made recently. But there are more processors. You can find all of them in the equipment section of the studio menu.

In the picture on the right you see a simplified version of my studio tech schematic again with only the relevant stuff on it for my effects equipment. I have a combination of analog and digital Sound Effect Processors. The analog processors are connected to one of the AD/DA converters either directly or through the SMP16 patch-bay. The digital Sound Effect Processors either have ADAT and are directly connected to the RME ADI-648 or they have SPDIF or AES/SBU and are connected to the Friend-Chip. The Friend-Chip converts the SPDIF or AES/SBU signals again into ADAT and is connected to the ADI-648 again. In every case the signal goes in two directions. The clean signal goes from the PC into the sound effects processor and the processed sound comes out again and is send back into the PC. And as said in the Midi Routing article the midi connections are used to synchronize the Effect Processors to the BPM of the Sonar Project through Midi Clock.

To be able to do this I use a special plug-in in Sonar called 'External Insert'. In this plug-in I can configure the input and output channel that the outboard Sound Effects Processor in connected to. When this is done you need to set the outboard gear into bypass mode so it does nothing but send back the signal it receives unaltered. Then you click a button on the plug-in and it measures the time it takes to receive the audio back that it sends out. It then uses this time to compensate the delay when you run audio from Sonar through this plug-in. And then you just save the effect you just configured as a preset and you are ready to go. I usually then setup a audio bus in Sonar so that I can send a certain amount of signal from an instrument track into that buss. And then I set the external outboard gear to 'full wet' so that it only sends back the processed signal without the original 'dry' signal.

I said before that I had some Sound Effects Processors that were connected directly to the RME ADI-648 with ADAT. You might wonder why they need 8 channels. Well some of them are capable of doing Surround Sound Processing even or do multiple effects at the same time. And there for it uses more than 2 channels that would be necessary for stereo processing. You can see two of these ADAT connected machines in the picture. On the left is the Eventide H8000 and on the right the Kurzweil KSP8. Both very powerful effect processors that can do surround, but usually I use the Eventide in a Dual Stereo setup and the Kurzweil even in a Quadruple Stereo setup. So the eventide runs 2 different algorithms at the same time and the Kurzweil does 4! It is great that these effects all are usable in Sonar through the External Insert Plug-in, just like you would use a software plug-in, but then you do get the sound of the outboard gear.

Apollo Studio Tech (Part 6)

Another very important protocol in my studio is Midi. Midi is used to record the notes you play on a synthesizer on the computer and then you can also play the notes back on the synthesizer. You can also use it to  play remotely on a synthesizer or send and receive data. It works a bit like a serial (RS-232) connection, but with separate cables with male DIN-5 connectors on both ends for  sending and receiving. I have a lot of synthesizers and that means a lot of Midi connections. Almost all of my synthesizers are connected with both Midi In and Midi Out to a dedicated port. The reason I do this is that in this way I can use all keyboards to play on and record in my sequencer and also program sounds on all synthesizers remotely through a program I use called 'Midi Quest'. This software can also backup sounds from the synthesizers and put new sounds in them with a click on a mouse button.

As you are used from my by now you see a simplified version of the studio schematic on the right with only the midi connections. As you can see I use several Midi interfaces. The actual number is even bigger, there are 5 of these midi interfaces connected to the Audio PC alone. They all have 8 inputs and 8 outputs. Some synthesizers also have USB connections and then they emulate midi over a USB. The midi interfaces I use are all from the Motu brand. I like them a lot. Very straight forward. The only problem I still have with them is that the order that Windows sees the interfaces in isn't the same every time I reboot my PC. This is quit annoying since I need to find out then which interface is which before I can start using a synthesizer connected to a certain port.

 To get even more ports I use Several Roland A880's. These are stand-alone Midi patch bays. They also have 8 inputs and outputs and with the buttons on the front you can determine which port is connected to which. And also you can store setups as a preset.  I have a A880 connected to every Midi interface. That gives me a total of 14 midi ports per midi interface, since on the interface itself I use port 8 to connect to port 1 of the A880. I also a Roland A220. This is a midi splitter. It has multiple outputs and basically I use it just to split up one midi output to several midi outputs, but I left it out of the drawing. I use it in my modular setup.

 I also use two Anatek SMP-16's. These are very special modules that are not only midi devices. It is an automated patch bay that  has both audio and midi inputs and outputs. I use them with my analog Sound Effects Processors to create flexible analog audio routings, but also to replicate the midi clock signal coming from my sequencer software to keep all the Sound Effect Processors to run in Sync.

Like my audio setup I tried to keep the midi interfaces as close as possible to my synthesizers and other midi enabled equipment, so I can use short Midi cables and transport USB to the PC over the longer distances.

03 August 2010

Apollo Studio Tech (Part 5)

I wrote in previous articles that all the digital audio equipment is running on 48 Khz in my studio. When you transfer digital signals at this rate it is very important that the timing of these signals is perfect to avoid loss of even a single one or zero. Normally when you connect two digital machines together it is possible to configure one as 'master' and the second one as 'slave'. The slave will synchronize automatically to the master. When you have a lot of digital equipment though it is not that simple anymore. Everything needs to be in perfect sync at the same time. For this purpose a special protocol was designed called 'Word Clock'. It is nothing more than a very steady 48 Khz signal that is send over a coax cable.

In the picture on the right you see again a simplified version of my studio schematic with only the word clock signals showing in purple. Again this is just a symbolic representation. The actual wiring will be a bit different, but you can see that the Lynx Aurora's, ADA-8000's , ADI-648's and Friend-Chip DMX 's are all connected to word clock. Also the Fireface 800 that is connected on my mastering PC is connected to Word Clock. Normally this PC is internally clocked on 44.1 Khz, but I can synchronize it to the Big Ben when I want. There are also a couple of Sound Effects Processors connected to World Clock actually, but I didn't put them in this schematic. But I guess you get the picture by now. You also see another piece of equipment I didn't talk about before that is called 'Big Ben'.

The Big Ben is made by Apogee and is simply a clock generator, but it is a very good one. It produces a very stable clock and has multiple outputs. In the past I used my PC as master clock but that had some disadvantages. To start with not everything would sync to it and it only had one output. The Big Ben has six outputs with build in terminators. For a stable clock signal and good distribution the coax cable that the signal is running over must be a 75 ohm (impedance) bus structure that needs to be terminated on both ends with 75 ohm terminators.

A terminator is nothing more than a 75 ohm resistor build in or attached to a BNC connector. These can be build in the equipment or mounted externally. In the picture on the right you see how a simple Word Clock bus would look like. You see two terminators on both ends. Special T or Y connectors that connect the equipment to the bus. One master device that in my case will be the Big Ben and two slave devices.

On the left you see a BNC cable with coax connectors attached. It is very important that you have the right cable and terminators. In the past the Ethernet network was also transported over coax, but this cable and also the terminators that came with them were 50 Ohm in stead of the necessary 75 Ohm. So don't use those!

I'm very happy with the Big Ben. One of the other nice things it does is measure the impedance on all its outputs. If one of the Word Clock buses is not exactly 75 Ohm a red led will light in stead of the green led for normal operation. So you can instantly see if everything is alright. All my equipment is in perfect sync now and the quality of the digital recording is outstanding because of this. A bad sync of equipment leads to dramatic decrease of clarity of the recordings immediately. I had a lot of trouble with this in the past.

02 August 2010

Apollo Studio Tech (Part 4)

Since I have a lot of digital audio signals I need to be able to creatively patch and merge signals. Again I made a stripped version of the studio schematic in the picture on the left. Here you see three boxes called 'DMX12', ''DMX16 and 'DMX32'. These are Friend-Chip digital patch bays / audio routers. The DMX32 and DMX16 are two very important pieces in my studio, because they are able to convert SPDIF  signals to ADAT and back. It can also patch an input signal to multiple outputs. And doing all this it can also convert the sample rate at the same time. As I said before my whole studio runs on 48 Khz. I try to put the output sample rate of my digital synthesizers on that sample rate where possible , but some synthesizers are only able to send out a 44.1 Khz SPDIF signal. The DMX32 and DMX16 are able to convert this to 48 khz. That sounds easier than it actually is. I'm sure Friend-Chip has to do a lot of magic for that.

On the right you see the DMX12 and DMX32 from the front and the back. These pictures are not mine, but come from the Friend-Chip website. The DMX12 is a standard product so it looks exactly the same as mine. It has 3 coax inputs and outputs and the rest is Toslink. It is able to patch SPDIF coax to Toslink, but it can only patch ADAT from Toslink to Toslink. No sample rate conversion on this box. So basically the DMX12 is just a automated patch bay and that is exactly what I use it for. It has a midi input and output as well so that you can configure it remotely and select presets.

The DMX32 and DMX16 are much smarter. They are chassis based and you can configure them like you want by inserting modules. The DMX32 is a 2HE unit that can hold up to 8 modules, where the DMX16 is a 1 HE unit that can hold 4 modules. In the DMX32 I  inserted one SPDIF coax module with 4 inputs and outputs and the rest is Toslink. In both of them are also 2 MAQ modules. These ones are able to fold 4 SPDIF signals (2 channels) into 1 ADAT channel (8 channels) and also the other way around. With these modules I can patch a digital synthesizer with SPDIF out directly into an ADAT channel on the ADI-648 MADI converter. I also have digital effect equipment attached to them. I can easily route signals from my audio PC through an Effects Processor and back or put a synthesizer through an effect unit before it goes into the ADAT channel. That makes my setup extremely flexible. Friend-Chip offers a nice Java based program that looks like a matrix. You can make patches just by the click of the mouse and save complex setups as presets and recall them later. They are controlled by Midi. I really love these boxes. You can find more information on the Friend-Chip website: http://www.friend-chip.de/

27 July 2010

Apollo Studio Tech (Part 3)

In this article I want to talk a bit about the AD/DA converters in my studio. On the left you see again a stripped down version of the studio schematic. You can click it again for a bigger version. You see the analog synthesizers on the left and the analog audio signals coming from them in red going into the AD/DA converters en coming out on the right in green as ADAT. There are two types of AD/DA converters in this picture marked 'ADA8000' and 'Aurora'. I will go more in dept on this equipment further on in this article. But what do these converters do? Well quite simply they convert analog signals to digital and digital signals to analog. As I stated before they all run on 48 Khz, meaning that 48.000 times a second they measure the voltage of the analog signal and translate that into digital numbers that are transferred in 'ones' and 'zeros'. In my setup I only use AD/DA converters that convert to from and to the 8 channel ADAT protocol. But also this could have been SPDIF for example.

In AD/DA converters there is a lot of quality difference. The best converters I have are two Lynx Aurora 16's. They have 16 mono input channels and 16 mono output channels on on the analog side and I inserted two ADAT boards so that is also has 2 ADAT input channels and two ADAT outputs. I use one of them in my analog corner, because especially analog synthesizer have a very wide dynamic range. These aurora's are able to truly capture the spirit of the analog signal without losing the dynamics. They sound terrible accurate. The other one is connected to some of my favorite synthesizers in my Digital Corner. On both I also use some outputs, but I bought them especially for the inputs. One of the things to keep in mind is that these Lynx Aurora converters become very hot. They really need ventilation on the top, so you cannot put them in the middle of a rack between other equipment. In both racks I have them on top for optimal ventilation. You can find more information on these converters on the Lynx Studio website at: http://www.lynxstudio.com/product_detail.asp?i=1

I also use a some Behringer ADA8000 AD/DA converters in my studio. These are much cheaper than the Auroras, but also a bit less in quality off course, though they are really not bad at all. I know a lot of people are prejudiced about Behringer, but they actually make some quite good stuff as well, like the whole 2496 series for example. These converters have only 8 channels in stead of 16 and there is also just one ADAT input and one ADAT output on board. They do a good job in my studio. I have a couple of them in my racks. I use them for synthesizers that I use less often and also for integration of Analog Sound Effects Processors.

I placed all the AD/DA converters as close to the my synthesizers as I possibly could. I also created several islands in the studio for that in the studio with a lower 19 inch rack, with a midi interface and an AD/DA converter in it. In this way I can use the shortest possible analog copper wires for the analog signals and from there I go on with digital signals. The advantage off course is less noise and hum, but also less cable spaghetti :) From the AD/DA converters to the rest of the digital equipment the signals are transported with ADAT over optical fiber. I have patch panel connected as well to every AD/DA converter so I can easily patch equipment to them. And off course everything is neatly labeled so I always know what is what. In the picture you see the Lynx Aurora in my Digital Corner.

25 July 2010

Apollo Studio Tech (Part 1)

A while ago I made a schematic like this for my previous AtmoSphere studio, but lots has changed since then. So last week I made a new schematic for my current Apollo Studio. You can see the main components in my studio and the connections between them. I used different colors to show the different protocols that are used. Please keep in mind that one line doesn't mean it is just one wire. It can represent more connections that all would do the same thing. It is just a logical layout. You can click the picture of the schematic for a larger version. I will post some more articles in the coming period to explain what it all it.

Here already some colors and what they represent:
  • Red - Analog audio
  • Blue - Digital audio (SPDIF or AES 2 channels of audio in one fiber)
  • Green - ADAT (8 channels of digital audio in one fiber)
  • Dark Yellow - MADI (64 channels of digital audio in one fiber pair)
  • Purple - Word Clock
  • Gray - Midi
  • Dark Magenta - USB
  • Dark Purple - Firewire
Some colors are a bit difficult to distinguish, but the articles coming will explain it all. What you can discover now is that the audio path in the schematic is going from left to right. Starting at the synthesizers and ending at the monitor speakers. And the midi and USB is running between the synthesizers, FX and the Midi interfaces and A880's. Well have fun looking at it and more is coming :)

15 May 2008

First go at Room EQ

Last week we had a first go at trying to improve the acoustics in my studio. As I stated before my friend Hanz heard some nasty peaks in the low end of the sound. I bought a Behringer FBQ2496 to solve these problems. The FBQ2496 is basically a very precise equalizer. You can set filters to lower the level of the problem frequencies.

We hooked up a Behringer measurement microphone to the fireface 800 on my mastering PC. This was the easiest way since this microphone needs phantom power. After that we downloaded the Room EQ Wizard software from: http://www.hometheatershack.com/roomeq. Surprisingly this software is free to use. And it is widely used by audiophiles and professionals around the world.

After setting up the software and adjusting the right position and input levels for the microphone, we did a couple of measurements. In the picture on the left you see the result. The blue line is the ideal situation, and the red line is the actual situation. As you can see we only focussed on the low end up to 200 Hertz.
As you can see there is a big dip around 100 Hertz and quite a big peek after that. This has to do with room resonances. Also between 40-50 and 80-90 there are some peeks. After measuring the software calculates the filters it has to set to compensate for the peaks. Unfortunately this is the only thing you can fix a bit. The dip is not fixable Hanz explained to me because it is caused by reflections that cancel each other out in the room. If you make them louder they still will cancel each other out and it will have the same result.

In the picture in the right you see some new lines. Again the same blue ideal line but now another light blue line is added. These are the actual filter calculations that the software did. One nice feature of room EQ is that you can transmit these values over midi to the FBQ2496. Unfortunately though this didn't seem to work. So we entered the values manually in the equalizer. Next time we will spend some more time figuring out why the midi transmit didn't work. The red dotted line you see in the picture now is the curve that Room EQ wizard now predicts to be actual in the room. We did not have time to measure again. So we will have another go somewhere in the future. We did do some listening tests with the FBQ2496 running and put it in bypass mode a couple of times and we could hear the difference. It was subtle but the low end definitely got more definition (less muddy). So the technical side works. Now we have to try to improve it a bit more. Off course the best way to solve these problems would be to treat the room with acoustic materials, but that is not going to happen anymore in this studio. I will be starting a very exiting new project that I will tell you about soon :)

03 May 2008

AtmoSphere Studio Tech Overview

Since a lot of people liked the series I did about the technical side of my current AtmoSphere studio, I decided to create one last post about this topic with an overview of all the previous articles. In this series I explained how everything works in my studio, beginning with the complete schematic overview of all the wiring you see here on the left and then leaving stuff out to make it easier to explain everything. The main reason I did this series is just that I hope it is helpful to others. Mainly because digital audio is a puzzle to many musicians I know. OK here are the articles:
So I hope you will enjoy this. If you have any questions, just send me an E-mail. I'd be happy to try and help you out.

05 April 2008

AtmoSphere Studio Tech (Part 9)

In the previous posts I focussed mainly on the input side of my audio PC, but off course it is also quite nice to hear the output somewhere. I work a lot at night in my studio so for a large part I rely on my headphones. Most mixing I do on a Sennheiser HD590. I really love the details you can hear on these. The only thing that is hard to judge on headphones is the low end (bass) of a mix. For that I use a surround setup of Behringer Thruths. They are not the best monitors you can buy, but they are very nice to just listen music on as well. So more Hifi graded. I still want to buy a decent pair of stereo monitors to do my mixing on, but I haven't heard anything that I really like yet.

I also use two monitor controllers in my studio. The most important one is the Presonus Central Station. I really love this one. You have two headphone inputs and it also has a very good AD converter build in. So I can directly put SPDIF in there. In the picture on the right you see again a simplified version of my studio schematic. You can see that two SPDIF signals go to the central station. The other Monitor controller is a SPL surround controller. My Behringer Thruths are connected on there. But I route the front stereo speakers from the surround setup throught the central station now temporary until I find new stereo monitors, but I will wait with that until my new studio is ready? Again a new studio??? Yes :) I have some nice new plans that I will unravel soon here on my blog. So keep watching this blog for news.

29 March 2008

AtmoSphere Studio Tech (Part 8)

In the past I used only software sounds effects like reverbs, filters and delays. While mastering my first album at Groove though I heard how much better their hardware sounded. So I started looking for some outboard gear as well. The advantage of outboard gear is off course that it is dedicated for that purpose only and it saved a lot of CPU usage on the audio PC. Disadvantage is that syncing is a bit more difficult and you have to compensate for latency. But sonar takes care of that for me. In the picture on the left you see a simplified version of my studio tech schematic again with only the relevant stuff on it for my effects equipment.

The two units I use the most in my studio are my reverb units. My best one is an Eventide Eclipse. The sound of this one is so great I cant describe it, everything you run through it becomes better :) It doesn't to only reverb, it also has delays and harmonizer patches. The second unit I use is a Behringer REV2496. It sounds very good and is really underrated. The brand Behringer has a bad name. This is because they sell some very bad products as well. But actually the REV2496 is very usable and it is even dual engine so you can use it twice. I use both units via SPDIF in and out connected to the Friend-Chip DMX32. This way I can route audio through it easily. Since I use them digitally both of they are also synced to Word Clock on the Big Ben. You can find more information here:

http://www.eventide.com/AudioDivision/Products/Harmonizers/Eclipse.aspx
http://www.behringer.com/REV2496/index.cfm?lang=ENG

I also use two units that are attached directly to the PC. The top ons is a Pod Pro XT. It is actually a guitar effects units but I use it for synthesizers. It has nice delays and distortion models. It is connected with USB to the audio PC. I use it with a software program called Gearworks that makes it available as a VST plugin in Sonar. I can use it multiple times and this way it syncs perfectly. The other unit is a PowerCore X8 that is attached with Firewire to my audio PC. It works also as VST's inside Sonar. For both units you can buy extra models of classic reverb and tape delay units. For the PowerCore I bought the TC Electronics System 6000 plugins and they sound really great. The nice thing about the VST stuff is off course that all settings are saved with my projects as well. You can find more information here:

http://line6.com/podxtpro/
http://www.tcelectronic.com/PowerCoreX8.asp

Next to all this high tech effect units I also use some low tech guitar effect units that I usually run one synthesizer trough directly. My favorite units are the Elektro Harmonix Mistress that you see on the left in the picture and the Elektro Harmonix Smalltone (Russian version). Both are phasers that were used by Jean-Michel Jarre on his famous Oxygene album. If I run my Eminent 310 organ (that Jarre also used on this album) through it you really get that genuine Oxygene sound. I love it. But these units are also very usable on other synthesizer to make the sound more alive. Besides this I also use some other units like the Boss SE-50, SE-70 and the RE-20 that I recently bought. More information:

http://www.ehx.com/
http://www.roland.com/products/en/RE-20/index.html

Using good Effect Equipment really makes your own sound. I'm always looking for more special stuff and found that especially guitar pedals make a nice addition. They are cheap and strange :)

21 March 2008

AtmoSphere Studio Tech (Part 7)

Another very important thing in my studio is the Midi protocol. Midi is used to transfer the notes you play on a synthesizer to the computer and also play remotely on a synthesizer. I have a lot of synthesizers and that means a lot of Midi connections. Almost all of my synthesizers are connected with both Midi-in and Midi-out to a dedicated port. The reason I do this is that in this way I can use all keyboards to play on and record in my sequencer and also program sounds on all synthesizers remotely through a program I use called 'Midi Quest'. This software can also backup sounds from the synthesizers and put new sounds in them with a click on a mouse button. Very convenient.

As you are used from my by now you see a simplified version of the studio schematic on the right. As you can see I use 5 Midi interfaces with 8 inputs and outputs each divided over two different computers. I didn't plan to do this, but the problem is that I also have a lot of synthesizers that emulate midi over a USB connection and all the drivers together wouldn't run on my main PC. So I decided to hook up all the modules in my rack to my mastering PC and use that as a Midi router. I use a little freeware program called 'Midi-OX' to make the connections. The midi interfaces I use are all from the Motu brand. I like them a lot. Very straight forward. I just hate Windows XP for not being able to support everything in the same time.

For the stuff I don't use Midi on a lot, like my FX equipment I use two Roland A880's. These are Midi patchers. They also have 8 inputs and outputs and with the buttons on the front you can determine which port is connected to which. And also you can store setups as a preset. I recently also bought a Roland UM-550 that is basicly a USB midi interface, but I use that one stand-alone as well as a midi patcher. The last Midi device I use is an Roland A220. This is a midi splitter. It has multiple outputs and basically I use it just to split up one midi output to several midi outputs.

Like my audio setup I tried to keep the midi interfaces as close as possible to my equipment, so I can use short Midi cables and transport USB to the PC over the longer distances. So far this setup works fine for me. But I would like to connect everything to one PC in the future. With Windows Vista it is possible. I already tested that, but unfortunately there is no support on Vista for a lot of my older equipment.

09 March 2008

AtmoSphere Studio Tech (Part 6)

I talked before about all the digital equipment running on 48 Khz in my new studio. When you transfer digital signals at this rate it is very important that the timing of these signals is perfect to avoid loss of even a single one or zero. Normally when you connect two digital machines together it is possible to configure one as 'master' and the second one as 'slave'. The slave will synchronize automatically to the master. When you have a lot of digital equipment though, like I do, it is not that simple anymore. Everything needs to be in perfect sync at the same time. For this purpose a special protocol was designed called 'Word Clock'. It is nothing more than a very steady 48 Khz signal that is send over a coax cable.

In the picture on the right you see again a simplified version of my studio schematic with only the word clock signals showing. Again this is just a symbolic representation. Actual wiring is a bit different, but you can see that the AD/DA convertors, MADI interface, Friend-Chip DMX and the digital mixer I talked about before all are connected to word clock. You also see another piece of equipment I didn't talk about before that is called 'Big Ben'.

The Big Ben is made by Apogee and is simply a clock generator, but it is a very clever one. It produces a very stable clock and has multiple outputs. In the past I used my PC as master clock but that had some disadvantages. To start with not everything would sync to it and it only had one output. The Big Ben has six outputs with build in terminators. For a stable clock signal and good distribution the coax cable that the signal is running over must be a 75 ohm (impedance) bus structure that needs to be terminated on both ends with 75 ohm terminators. A terminator is nothing more than a 75 ohm resister build in or attached to a BNC connector. These can be build in the equipment or mounted externally. In the picture on the right you see how a simple Word Clock bus would look like. You see two terminators on both ends. Special 'T' connectors that connect the equipment to the bus. One master device that in my case will be the Big Ben and two slave devices.

On the left you see a BNC cable with coax connectors attached. It is very important that you have the right cable and terminators. In the past the ethernet network was also transported over coax, but this cable and also the terminators that came with them were 50 Ohm in stead of the necessary 75 Ohm. So don't use those!

I'm very happy with the Big Ben. One of the other nice things it does is measure the impedance on all its outputs. If one of the Word Clock buses is not exactly 75 Ohm a red led will light in stead of the green led for normal operation. So you can instantly see if everything is alright. All my equipment is in perfect sync now and the quality of the digital recording is outstanding through this. A bad sync of equipment leads to dramatic decrease of clarity of the recordings immediately. I had a lot of trouble with this in the past.

01 March 2008

AtmoSphere Studio Tech (Part 5)

Since I have a lot of digital audio signals I need to be able to creatively patch and merge signals. Again I made a stripped version of the studio schematic in the picture on the left. Here you see two boxes called 'DMX12' and 'DMX32'. These are Friend-Chip digital patch bays / audio routers. The DMX32 is another very important piece in my studio. It is able to convert SPDIF to ADAT and back. It can also patch an input signal to multiple outputs. And doing all this it can also convert the sample rate at the same time. As I said before my whole studio runs on 48 Khz. I try to put the output sample rate of my digital synthesizers on that sample rate where possible , but some synthesizers are only able to send out 44,1 Khz. The DMX32 is able to set this all straight. That sounds easier than it actually is. I'm sure Friend-Chip has to do a lot of magic for that.

On the right you see the DMX12 and DMX32 from the front and the back. These pictures are not mine, but come from the Friend-Chip website. The DMX12 is a standard product so it looks exactly the same as mine. It has 3 coax inputs and outputs and the rest is Toslink. It is able to patch SPDIF coax to Toslink, but it can only patch ADAT from Toslink to Toslink. No samplerate conversion on this box. So basically the DMX12 is just a automated patch bay and that is exactly what I use it for. It has a midi input and output as well so that you can configure it remotely and select presets.

The DMX32 as is much smarter. It is chassis based and you can add configure it like you want with modules. I inserted just one SPDIF coax board with 4 inputs and outputs and the rest Toslink. There are also 2 MAQ interfaces in it. These ones are able to fold 4 SPDIF signales (4 x 2 channels) into 1 ADAT channel (1 x 8 channels) and also the other way around. With this module I can patch a digital synthesizer with SPDIF out directly into an ADAT channel on the MADI interface. I also have digital effect equipment attached to it so I can easily route signals from my audio PC through it or put a synthesizer through an effect unit before it goes into the ADAT channel. That makes my setup extremely flexible. Friend-Chip offers a nice Java based program that looks like a matrix. You can make patches just by the click of the mouse and save complex setups as presets and recall them later. Like the DMX12 this is controlled by Midi. I really love these boxes. You can find more information on the Friend-Chip website: http://www.friend-chip.de/

24 February 2008

AtmoSphere Studio Tech (Part 4)

In this posting I want to talk a bit about the AD/DA converters in my studio. On the left you see again a stripped down version of the studio schematic. You see the analog synthesizers on the left and the analog audio signals coming from them in red going into the AD/DA convertors en coming out on the right in green as ADAT. There are three types of AD/DA converters in this picture marked 'AD8000', 'DDX3216' and 'Aurora'. I will go more in dept on this equipment further on in this posting. But what do these converters do? Well quite simply they convert analog signals to digital and digital signals to analog. As I stated before they all run on 48 Khz, meaning that 48.000 times a second they measure the voltage of the analog signal and translate that into digital numbers that are transferred in 'ones' and 'zeros'. In my setup I only use AD/DA converters that convert to from and to the 8 channel ADAT protocol. But also this could have been SPDIF for example.

In AD/DA converters there is a lot of quality difference. The best converters I have are two Lynx Aurora 16's. They have 16 mono input channels and 16 mono output channels on on the analog side and I inserted two ADAT boards so that is also has 2 ADAT input channels and two ADAT outputs. I use of in my analog corner, because especially analog synthesizer have a very wide dynamic range. These aurora's are able to truly capture the spirit of the analog signal without losing the dynamics. They sound terrible accurate. The other one is connected to some of my favorite synthesizers in my Synth Alley. On both I also use some outputs, but I bought them especially for the inputs. One of the things to keep in mind is that these converters become very warm. They need ventilation on the top, so you cannot put them in the middle of a rack between other equipment. In both racks I have them on top for optimal ventilation. You can find more information on these convertors on the Lynx Studio website at: http://www.lynxstudio.com/product_detail.asp?i=1

I also use a some Behringer ADA8000 AD/DA converters. These are much cheaper than the Aurora, but also a bit less in quality off course, though they are really not bad at all. They have 8 channels in stead of 16 and there is one ADAT input and one ADAT output on board. They do a good job in my studio. I have two of them in my main rack and they are connected to a Behringer DDX3216.

The DDX3216 is a very compact 32 channel digital mixer that I use to connect all the synthesizer modules in my rack to. It has 16 analog input, so in this way it acts as a AD converter on itself as well. I also inserted two double ADAT interfaces, so also it has 4 ADAT inputs and 4 ADAT outputs as well. The routing in this mixer is very flexible so actually I use it only as sort of a digital patch panel and AD converter. I can easily say which channel is going to which bus and the buses are routed to the ADAT outputs. Off course it is easy to set the input level on the mixer, but actual mixing is done in my Audio PC. This mixer sounds very transparent and that is what I want from it. I don't like coloring of my audio signals. The DDX3215 has some build in effects as well, but I use none of these, because I think they are not so good.

I placed all the AD/DA converters as close to the analog synthesizers as I possibly could. In this way I can use also the shortest possible analog copper wires. The advantage off course is less noise and hum, but also less cable spaghetti :) From the AD/DA converters to the rest of the digital equipment the signals are transported with ADAT over fiber. This saved me a lot of cabling in my current setup as well. I have a patch panel connected as well to every AD/DA converter so I can easily patch equipment to them. And off course everything is neatly labeled so I always know what is what. I'm very happy that I invested in this setup since it saved be about three weeks building up my studio again I guess and besides the overall quality of the sound has improved dramatically. But also within a year I hope to move everything again to an even newer studio :) Curious? Keep track of my blog. I will tell more about that soon.