Glenn Stanton

in addition, i've also used the Cakewalk Session Drummer and a number of the drum programs come with a variety of percussion - some is in WAV, others FLAC, but overall decent enough where @bitflipper and @John Vere noted percussion isn't front and center. other times i use the AD 2 session percussion when something a bit more real is needed (although not a fan of the congas, so i'll either play those myself or record as a sample).

btw, a workaround i've used is the comping features to create separate tracks. i've always been a bit nervous about long recordings when transferring from tape. so i'll loop on say 100 measures @ 60bpm, then start recording (this is how i've been extracting several hours of tape recordings into the digital realm). then each track can be readily rejoined as needed. typically i'm splitting on between song sections then creating a separate new track for each song and bounding to create individual clips. so far i've managed about 250 hours without an issue. oh, make sure you're doing this to a fresh disk since it will eat your regular disks up pretty quickly. since i don't care about the maximum quality - i'm using 44.1@16bit - i'm using an inexpensive 1TB HD.

agreed, if this is doing it on two different systems and cables, you probably have a defective IO unit. may use another device or borrow one to test that it goes away. perhaps even the internal audio device might be enough to complete the assessment.

hmmmm. is the networking off or disabled? including bluetooth etc? this tends to be the biggest offender when there are audio system interrupts involved. in most cases i leave my networking on but turn off or pause my OneDrive and Google Drive apps. i don't use bluetooth so that's disabled, as we ll as my other audio sources like webcam and internal audio.

thanks! i don't usually have problems except for the 2x per year for massive updates ? more likely it's when i have other software products which have 2 or more runtime libraries to install that somehow squash my drivers. this is good info though.

as a simple principle - whenever Microsoft does a large update on Windows, or i get issues like this after install some software, i just re-install my drivers and reboot. fairly painless (10 mins or so) but seems that the "stacking order" of runtime libraries gets confused somehow, so i make sure my drivers are "last in - wins"... and so far for the past 2 years it's been a reliable strategy...

are your monitors functioning when you play an MP3 file or something from YouTube? if not, step one is to make sure the speakers are connected to your outputs on the UMC interface, levels are up enough to hear sounds, and any "USB <> Direct" knob is set to USB. make sure you're getting sounds from Windows before setting up CbB. you need to go to the preferences section and make sure your audio device is selected for both inputs and outputs. if you don't see in the device list, you need to make sure the ASIO setting is selected under the playback and recording section, then select the device. it's a good idea to check settings and make sure you have 24 bit selected in the driver settings section. close the preferences. if you run your demo project, you should hear it. if not: in your project, you need to make sure either the track(s) or master bus are set to use the device output. and when recording, the track needs to have its input set to the device channel (left, right, stereo) you have your instrument or mic plugged into. arming the track let's you see the input meter operate so you can adjust your input level on the interface to put the meter at -18db to -12db peaks.

i had similar problems with the version 4 product (Editor) but the upgrade to 5 fixed it for me (as well as the newer features). one thing that happens on occasion is the timer circle completes but the screen never seems to refresh, so i have to close the tab then re-open. this only seems to happen on long segments with a lot of detail (like if i'm converting an acoustic guitar track to midi and there's a lot of notes to analyze. time keeping wise - it's in sync for me. have not seen the double-spacebar thing except when i'm being spastic trying to get playback to stop or start when i'm using an older 32-bit plugin ?

you have to check the ASIO4ALL control panel - mine sometimes slips into using the Realtek audio or the webcam instead of the mixer, but opening the CP and touching the USB (mixer) enables it.

@Starship Krupa if i just select the track (in this case BASS2) and bounce to track, nada. if i select both the KICK and the BASS2 track, then open Bounce to Track(s) dialog and then DESELECT the KICK, it works! so maybe you need to select the triggering track and the target track, then deselect the trigger in the BtT dialog... i also turn off fast bounce (in case the effects need processing time - based on help file - but experiment...)

i think there are sites where musicians often share their "secret" mix of effects, string sizes, vibrato/or/not, etc to get their sounds. i often watch these videos: https://www.premierguitar.com/topics/820-guitarists it's sometimes amazing how simple the rigs are for the most famous players, and of course some are also quite complex at times. some distortion, reverb, delays, maybe some chorus/phase/flange, and voilà! then you watch Edge, Andy Summers, Robert Fripp, and many more, and it's mind blowing the layering these guys use.

i answer them with "Acme Computers, please give me your name, telephone number, and a credit card number and someone will be right with you" then when they get confused, i'll say" please hold" and put them on hold, it's their dime on the call ?

@Starship Krupa i use a sidechain compressor on VI bass triggered by the kick all the time. just can't see it/hear it while soloing busses. i can solo the BOTH tracks and hear it though. you see in the graphics the kick is triggering the bass track compressor - both tracks need to be soloed for this to work or solo turned off across the board. bass is non-frozen: with the bass track frozen (kick -> bass sidechain): so it looks like it's being captured correctly.

MP3 is a very lossy audio format so all kinds of things get altered while it tries to "demask" in order to compress the file size. 320Kbs - not bad, 128Kbs - bad. so best bet - always export as WAV at recorded sample rate (don't up-sample, it's generally not useful, and if running 96K, skip down-sample until re-processing) + 24-bit and then use that file to generate the 41.1K + 16-bit MP3 file(s) (some online sites, or emails, limit the size or bit rate so its nice to have options w/o re-exporting). i tend to go with 48K+24-bit WAV to generate a 256Kbs MP3 since the file sizes tend to be just under the limit for reverbnation and other sites.

i've found even when the product says it requires Kontakt Full, it will run as a demo mode in player so every 12-15 min of playing time you need to restart Kontakt.

@John Vere the WASAPI works with my mixer - all modes do - and generally i run with the WASAPI Exclusive mode. but since i had the ASIO4ALL installed (my Alesis IO2 won't work with the WASPI and needs ASIO4ALL, but recently had one of the channels die), i figured i would run some tests. ? and in practice, i simply do a quick alignment if i find a track is offset. takes only a few moments. it's only live parts that typically need this when i'm running my remote rig. at home with the UMC1820 + ASIO drivers - it's pretty accurate all the time.

i have several "stem" busses which all feed a master buss. i only export by selecting only the master buss when i want the single combined output, or i can export all of the stem busses (low, vocals, instruments, solos, drums) in a single go by simply selecting those busses.

hey @Noel Borthwick here's the exact testing setup. note i'm using the laptop rig shown in my sig and profile. the device is a Behringer Q802USB mixer - limited to 16-bit and running at 48Khz. the only changes are the drivers selected: ASIO4ALL (there is no specific ASIO driver for this device) WASAPI Exclusive (WASAPI E) WASAPI Shared (WASAPI S) MME (32-bit) WDM/KS the "defaults" provided by each selection (ASIO = 512 samples, WASAPI 480, etc) all roughly 10-12ms loopback USB Codec (the default clever name...) Ctrl Out Left -> Ch4 Left In note: the second ASIO is the MAGIX Low Latency (ha ha) driver which is the (same as the?) ASIO4ALL driver under another name, and i chose not to use that one since in testing earlier it was identical to the ASIO4ALL results. as my previous note, when i select the 512 samples in the RTL under the Windows Audio Exclusive it glitches, but works fine on the 480 - hence my suggestion there is a bug in the RTL (still a beta product...) CbB does not glitch until i set the slider (on any of the drivers) to 32 or less samples ? (that's good and it's working well). however, there is definitely some delay compensation need with this setup. i'll be home in a week or two and i'll try to get my home rig up and running enough to test (although i'm in the porocess of moving so it may take longer). the home rig is the UMC1820 with ASIO drivers (and generally has always worked as expected).

@Noel Borthwick RTL for the WASAPI (Exclusive) was 2384. manual offsets didn't change the offsets in the recordings. if i changed it via the slider, then there was a small change - see pics. it's interesting that the RTL see 480 samples but the driver appears to be set using the previous 512 samples of the ASIO driver. when trying this using the RTL 512 sample setting it glitches out saying too short, increase samples, but going back to 480 then works fine... may be a bug there... no difference setting the manual offset move slider (bottom track):

yes, the loopback testing with RTL showed up as longer (1968) than the reported amount to CbB (1568) by ASIO4ALL. i noticed that the ASIO4ALL and MME were pretty close in terms latency, only off by the 32 samples shown in the ASIO4ALL control panel.

and if it's still problematic, run the installer locally as administrator.

i found on one of my USB interfaces, that it liked the longer older USB (steel braid shielded, heavy duty) over the short one supplied with it. we had someone here in the past 6 months who experienced an issue with their USB cable being old and too long (...no jokes please...) and replacing it with the newer heavier one "fixed" the problem. i'm guessing that as these devices get faster and more powerful (higher current levels for example in USB3) that the capacitance and induction of the wiring etc can impact it's sensitivity to EMI and race conditions (...the electronic kind...).

i just look at the big red button up top - if it's dimmed, nothing is armed, if it is lit up, then i have one or more tracks armed. in the latter case, i push it and it disarms all the tracks, on the other, if i press record and nothing happens, it's a "Doh!" moment and i try not to repeat it too many times before arming a track ?

the hmmm wasn't to condescend, i fully empathize with your situation. it was more a pause to consider that in the year since it went unsupport that perhaps some of the libraries which are required may have been updated or in some cases as i noted, confused in their stacking which has caused crashing or inoperation on my systems as well. cheers and hopefully this will be resolved.

i did some testing on my "away" system - laptop, Q802USB (16bit 48K), and using ASIO4ALL (only ASIO for this older device), MME, WASAPI Exclusive/ Shared, and WDM/KS. not surprising that the MME and ASIO4ALL are only about 32 samples apart (same as the ASIO control panel latency settings of 32 samples). on my system the WDM was closest with WASAPI (either mode) showing the worst. note - PDC was engaged (no plugins anywhere), ASIO reported 1568 samples. RTL reported 1968 with the loopback test on ASIO, 2231 for Windows Sound Exclusive, 7000+ for Direct Sound and Windows Sound Shared.