Hi, I am an older guy who hasn't really done much composing since the Cakewalk for DOS days using a Roland SCC-1. I tried a windows version of Cakewalk around 1998 but gave up on it. Fast forward to 2015 and I am using Win 7 64bit, and I'm perplexed and disheartened by the fact that I cannot figure out how to get my new Music Creator 7 software to work with my new Roland Duo-Capture EX (which I bought specifically to minimize any potential problems). When I open my old midi files in Music Creator 7 the only way I can actually HEAR something is to select Microsoft GS wavetable synth as the output. If I select Duo-Capture EX I hear nothing. I know it's working because I can use it to record midi, plus I can hear the Synth I use through an audio input. Now I've always used a sound module before to generate sounds, in fact I still have one, but the music store guy assured me repeatedly that I would be able to use 'soft synths' instead, I don't think Microsoft GS wavetable synth really counts. How do I access these great sounds I bought it for?

Deselect the Microsoft GS Wavetable synth. When no MIDI output device is specified, MC7 will use TTS-1, the bundled GM synth, when opening MIDI files.

and once you get used to using TTS-1 as a softsynth, explore other possibilities with the better sounds from Cakewalk SoundCenter synth.  it is NOT Multi-timbral like TTS-1 is, so that means you have to insert an instance of Soundcenter for each separate MIDI track and on each one you have to choose a specific sound for that instance of Soundcenter and load it into soundcenter.  but the benefits of doing all that will far outweigh the costs when you hear the differences between soundcenter and TTS-1.

Welcome to the form Nick.
 
Don't Panic.  Once you get the pieces figured out you won't regret your decision or your choices.
 
There are a lot more options than the (cheesy) GS Wavetable Synth, and the sound is sooo much better.  The trade off of having a lot better choices is that you have to make choices (and set things up).
 
My way is to load the MIDI into a MIDI track.  I then insert a soft-synth track, using any one of a lot of good synths.  The synth track is where the audio is actually created, so it needs to have its output set to your sound card (or to a bus that sends to the sound card when you start getting more complicated).  With the synth track in place, you need to choose the patch for the synth.
 
Direct the output of the MIDI track to the synth track.  Press play on the transport and you should be started.
 
The TTS-1 synth is a good one to start with, but there are a lot better sounding ones (Sound Center, SI) in MC7 once you get the basics down.
 
Glen
 
 

Switch to ASIO driver mode.
In the old days, you had to be running MME driver in order to use the MS Wavetable synth. I don't know if that's still true, but if it is, the Roland may not have an MME driver, thus no sound. The Roland will work better using their ASIO driver, I think. Go to their web site and download and install the latest ASIO driver for your device and OS if you haven't already.
And welcome to the forum.

Thanks for your kind replies. I wish I could post screenshots! If I open the browser window, under the Synth tab there is nothing so I add a synth a TTS-11, a dialogue box which I do not understand in the slightest, pops up. I click create and a new track with TTS-11 appears. From that point on the TTS-11 appears in the track output drop-down-menu along with Duo Capture EX and MS Wavetable synth, if I select TTS-11 I get sound, better sound than the MS Wavetable synth thankfully, but there is nothing coming out of the Duo Capture EX, I know this as it's on a different circuit than the computers speaker output, which is optical.
 
Okay, I go to the empty TTS-11 to record some new midi stuff, I choose Duo Capture EX as the track input and pick an instrument like saxophone. I arm the track and start recording, first there is incredible lag that makes this exercise virtually useless, there is a delay of almost a second between the time I play a note and it sounds through the software. Not only that, the instrument I've chosen, saxophone, isn't there, it's strings! I try another setting on my input device, a Roland AX-09 and I get piano. No saxophone, incredible lag, I'm stumped. 

Make sure you are running ASIO mode driver with the latest drivers from http://www.rolandus.com/products/duo-capture_ex/downloads/. To check the driver mode setting type P or from the Edit menu select Preferences. Then go into Playback and Recording. The driver mode drop down is at the top. Set it to ASIO. The ASIO buffer can be adjusted in Preferences > Driver Settings by clicking the ASIO Panel. This should display the Roland software to adjust the size of the ASIO buffer. The smaller the buffer the smaller the delay.

WRT getting your particular hardware synth setup I am going to have to leave that to a keyboard player.

Nick,
Are you using an external synthesizer?  How are you entering your MIDI?  Are you recording the MIDI with a keyboard or are you entering the notes using the PRV/Staff View?
 
Your Duo-Capture is really two devices.  One is a MIDI interface with a MIDI IN and a MIDI out.  Selecting the Duo-Capture as the output on a MIDI track is telling MC7 to send the MIDI notes to an external synthesizer that is plugged into the Duo-Capture.
 
If you are selecting the Duo-Capture on an audio track, it is sending sound to the audio interface on the Duo-Capture, which you would then hear through a set of headphones plugged into the Duo-Capture or through an external amplifier that is plugged into the Duo-Capture's L and R jacks on the back.
 
It sounds to me like you are trying to listen to the sounds using your computer's sound card (computers speaker output, which is optical).  This really defeats the reason that you want to use a good interface like the Duo-Capture.  You need to change over to the ASIO driver (Edit->Preferences->Audio->Plaback and Recording) and listen to the sound coming out of the L/R Jacks on the back of the Duo-Capture or the Duo-Capture's Phones jack.
 
The "delay of almost a second between the time I play a note" is called latency.  This will be bad until you switch over to ASIO and get your sound from the Duo-Capture interface.
 
Glen

There is a lot of work done to convert analog sound into digital data, which is what happens when you run audio output from an external keyboard into your computer.  That process is called Analog/Digital conversion, or A/D conversion.  The digital data would get stored on the computer as ones and zeros, rather than as actual sound.
 
When you load up a wave file on your computer, into a music program that can play that file, such as Windows Media Player, all of that digital data has to get converted back into audio data, from the digital data stored into the file.  That process is called Digital to Analog conversion, or D/A conversion.
 
For a simple song file, your computer can handle doing the work of these conversions, but if you are working with multiple tracks of data, it will become too much work for your computer's CPU to handle doing all of that conversion in real-time, and you end up with noisy crackly horrible sound, even with dropouts, because the CPU is not really designed to have to do that much work in real-time.  After all, it is also doing all the work of running the OS, and any programs you are running, etc.
 
Sooooo, the concept of transferring all of the conversion work A/D D/A was built into what we call an Audio Interface.  An audio interface has circuitry and software drivers that are purpose-built to handle all of the conversion back and forth from analog and digital, removing the need for you CPU to have to do that.  These audio interfaces are a critical component of a digital audio workstation (DAW) for running software like Sonar.
 
An audio interface with good quality converters and pre-amps and such can run around $150 on up to $500 or more, depending on features.
 
So, with an audio interface plugged into your computer, there are also software drivers needed so that the computer can talk to and control the audio interface.  The drivers will take on the work of telling the audio interface to convert a song file back to analog sound that can go to speakers connected to the audio interface, and those same drivers also would tell the interface to convert any data from inputs connected to the interface from analog to digital, and then pass that digital data to whatever music program you are using with the audio interface.  So, for all of the conversions in both directions, your CPU is spared.
 
So to get Windows to have your audio interface DO all of that work, you need to tell your music program to use the drivers for your audio interface - which will probably by a driver mode of 'ASIO'.
 
On MY computer, I have programs like Sonar, or stand-alone versions of Dimension Pro or Kontakt, all set to use the ASIO drivers of my audio interface, but I ONLY allow those kinds of programs to be accessing the ASIO drivers of the audio interface.  I have my Windows Default Audio Device set to use the on-board sound drivers of the computer's motherboard, and NOT to use the ASIO drivers of the audio interface.  I do this because of the way ASIO works - I let Windows Media Player, or songs played through YouTube, or other internet pages with audio clips all get handled by the computer's on-board sound.  Trying to share ASIO between Sonar and Windows Media Player or YouTube or whatever, leads to conflicts and just doesn't work.
 
So, for general settings for using an audio interface and ASIO drivers in Sonar, I do recording with the following settings: for the audio interface - Sample Rate 48 K, ASIO Buffer Size of 128.  For Sonar, settings for recording are: Driver Mode = ASIO, Sample Rate of 48 K, Record Bit Depth of 24 bits.  With these settings, I get a great quality of sound recorded, and low latency - meaning do discernible delay between singing or playing a note, and hearing it back through input echo on the track in Sonar.
 
Something you are going to want to do AFTER you finish recording, and are ready to load up a bunch of heavy duty effects, is to alter the settings.  This is because once you are mixing or mastering, you don't care about latency any more - you need to have a large enough ASIO Buffer so that things like a convoluted reverb can their look-ahead processing.  So, for mixing/mastering, I change my settings to: ASIO Buffer Size 1024 or even 2048, leaving the other settings the way they were.  That larger buffer adds a bunch of latency, but for mixing/mastering who cares?
 
For the rest of time, you will find yourself adjusting that ASIO Buffer Size back and forth, small (128) for recording, and large for mixing/mastering (1024 or 2048).
 
OK, so give the above a try and post back with your results.  :)
 
Bob Bone