TROUBLESHOOTING POPS IN RECORDED SAMPLES
If the samples you record with SampleRobot end up having unexpected pops in them, that's a sign your computer was running out of sample buffer while recording. If you open up the samples in an editor like Audacity or WaveLab
, you'll probably see something like this:
In my own work, I've only experienced this issue when using the virtual cable methods (and more often when using ASIO4ALL instead of just using VB-Cable directly). It could potentially happen when doing the analog interface-to-interface method, though. Here are some things to try when you run into this issue:
- The VB-Cable download package includes a control panel utility, even though the control panel is not actually installed on your system when you install the driver. When you extract the ZIP file that you downloaded from VB-Audio, you'll find the VBCABLE_ControlPanel.exe file inside. Just run it from the extracted folder.In the Options menu you can choose different "latency" settings (really you're just selecting a sample buffer size), and you can also adjust VB-Cable's internal sample rate. This sample rate is different from the input/output rates of program audio that you may be passing through the virtual cable. I'd recommend leaving both of these values at the maximum (7168 samples, 96000 Hz), although if you've tried all other tips below and you're still having problems you might try reducing the Internal SR (sample rate) setting to something closer to your recording rate. This will put less of a burden on your system and give you more mileage with the current sample buffer.
- Generally you also want to make sure that you've both been playing out of your instrument and recording in SampleRobot at the same sample rate and bit depth. Check your instrument/plugin host settings to make sure they sync up with what SampleRobot is expecting. Also, you may have noticed in my screenshot above that I had my VB-Cable set to 24-bit for input. This is due to a setting I made in the Sound control panel of Windows:
- If you are using ASIO4ALL, remember that sample buffers are device-specific. You have to actually have the specific device selected before adjusting the buffer size slider. With ASIO4ALL I find I pretty much always have to crank this up to the max:
- If you're using a standalone version of an instrument that also has a plugin version, you might want to try using the plugin version of the instrument in VSTHost instead. VSTHost has the option to adjust its sample buffer, which might give you enough breathing room to avoid audio glitches. See Part 1 of this series for some more information on VSTHost.
The VSTHost Wave Devices dialog has a sample buffer setting that many other hosts and standalone instruments don't have..
GETTING SAMPLE START AND STOP TIMES RIGHT
By default, SampleRobot auto-detects "note in" and "note out" times in your initial sample recordings so that when you export your sounds, they start right when an actual audible signal starts, and only run as long as the audible content plays. (In other words, SampleRobot may record an 8-second long recording of a 1-second drum hit, but when you export the sound with autodetection enabled, the exported sample will only contain that 1 second of audible audio.)The default settings usually work just fine for the VB-Cable methods I described earlier, and for the Digital version of the interface-to-interface method, but I've found that sometimes it's not all that accurate when doing the Analog method, likely because of the tiny amount of noisefloor that's almost always present in analog recordings.
In the Multi-Sample RECORD Settings portion of SampleRobot, you will find the Thres.Prec.In and Thres.Prec.Out settings. These affect how sensitive SampleRobot is to the differences in the audio signal between silence and audible sound. If you find that SampleRobot isn't setting the sample start times properly in your exported sounds, try adjusting the Thres.Prec.In value to something lower than the default value of 0.90. If SampleRobot is ending your exported samples before the audible sound has finished (especially common on reverb tails), try adjusting the Thres.Prec.Out value with something lower than the default of 0.50.
Note that you have to change these settings BEFORE you record a multi-sample. So this means you might have to go through several passes of recordings as you experiment with different values. As you do this, it's REALLY important to remember that SampleRobot's default behavior is to skip any samples that have already been recorded. So remember to disable this option if you have to re-record some samples:
If the auto-detection simply isn't working out perfectly for you, you can always manually tweak the note-in/note-out points on a per-sample basis before exporting. On the virtual keyboard on the bottom of the SampleRobot window, right-click the sample you'd like to adjust. A little "E" will appear under that sample in the virtual keyboard, and your sample's waveform will appear in the Note/Loop/Release Editor. It's a tiny window, but you can zoom horizontally and vertically by right-dragging your mouse in the editor window, and you can pan the view backward and forward by left-dragging in the view. The N. IN and N. OUT sliders set the start and end points, respectively.
, and everything's back to normal.
graphics card
to run two monitors. The drivers for the NVIDIA card I had before were
conflicting with the drivers for my PCIe audio interface on both
Windows 7 and Windows 8. This card is actually a downgrade in terms of
graphics power, but it uses less electricity and is quieter than the
NVIDIA card, and I hardly ever play games these days. It has both HDMI and DVI outputs, which is perfect for me since one of my monitors is HDMI only and the other is DVI/VGA.
keyboard controller. I thought the AKAI was a fine device, but it was
enormous due to all of its pads and faders and other controls which I
simply never used. I traded the AKAI's enormous feature set for the
Roland's light, slim design to reclaim some serious desk space.
, as it is pretty much the pinnacle of prosumer interfaces, fully intending to sell it off after my tests were complete. Problem is, the thing is so great, what with all the I/O options and the awesome TotalMix routing software, that I couldn't bear to part with it. Anyway, that's my current interface of choice. I'm using it via FireWire, but it works just fine via USB as well. The UFX replaces my 
, which is still a great, solid performer.
in one of my racks. Each bay has 24 in/out columns in a half-normalled configuration. (You can invert the individual boards to disable normalling.) This model also supports TRS/balanced connections, so I have to be careful which kind of cables I'm using when connecting up older synths with unbalanced outputs. (There are various recommended ways to do this, but I'm just making sure to use unbalanced interconnect and patch cables with them, and balanced cables from the patch bay to my interface.)
for most DAW controlling needs. (It works with other Steinberg products like 
too.) It integrates wonderfully with Cubase, and is pretty much a must-have in my day-to-day use, now- especially when tracking vocals. The bad news is that it is Steinberg-specific, and can't be
programmed to work in third-party DAWs.
. I got this because I wanted a physical device with transport controls
for Ableton Live. It has built-in templates to integrate with all major
DAW packages, and also includes buttons for setting and navigating
markers/locators in audio projects, plus 8 rows of channel strips which
map to tracks inside your DAW. A really handy little controller, at an
unbeatable price. Even though I'm mostly using Cubase these days, I still have this little guy wired up because it makes my life easier whenever I happen to be working in Live.
gives me all of this and more. You can switch between two different monitor sets, and even switch between two sets of inputs (one balanced, one unbalanced), and it's got a stereo/mono toggle for checking your mixes. This thing's big and weird looking, but it sounds great and has everything I need. I've heard some folks complain about the clicky master volume encoder. Well, the M-Patch even comes with a traditional pot-type volume knob that you can pop in if you don't like the encoder.
active monitors. But then the tweeters blew on my JBL passives, and now the E5s ARE my reference monitors! This things are pretty nice, although I will probably pick up some 
or 
monitors in the coming year, for a flatter response and lower noise floor.
to round out my sound. I don't have much experience with woofers, but the way this one's designed, you route sound from your audio interface's stereo outs to the subwoofer, and then from the sub to each of your active monitors. You set the crossover frequency on the subwoofer, so the monitors only receive the mid/high frequencies. It works really nicely, and has actually improved my mixing a bit. I discovered I had been mixing in too much bass before when I was working with the old setup. 
. I've been a cubase user since the very first
release on Mac in 1994, although the Cubase of today is wildly
different from the MIDI-only Cubase of the early 90s. After some
tweaking I seem to have 64-bit Cubase 7 running pretty stably. Version
6.5 seemed a little bit more reliable, but I'm doing okay for now. I believe that if you buy retail Cubase 7 now you get a free upgrade to the latest version, 7.5.
. With version 9, Ableton Live added some really interesting new
features and significantly improved a number of their built-in effects.
I've completed several productions in Live 9 and have found it to be a
really enjoyable environment. Both Live 8 and 9 offer 64-bit versions
now, too! I just really wish Live had VST 3 support; a number of plugin
developers only offer 64-bit plugins in VST 3 format, so Live's VST
compatibility isn't the greatest.
packages (especially 
). They really cover all the basics for software synthesis, and include a growing number of solid effects, too.
displays more information on your screen depending on your monitor's resolution. If you have a relatively low screen resolution, you might not be seeing all the default elements of your given workspace.
, which now shows the tab groups at its native resolution of 1920x1080.
. This device is fully supported in WaveLab for both Mac and PC, but on both OSes you need to have both 
, 
, 
, and 
. I imagine the mappings match what Steinberg did for the CC121, so the built-in help documentation for the CC121 is probably worth a read if you have the CMC controllers.
