Glenn Stanton

@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.

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 ?

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.

Windows does not have a native ASIO driver. what IO are you using? did you install the product ASIO drivers? if you recently did an update, did you reinstall the drivers? as a note, i find myself periodically reinstalling some of my drivers after a large Windows update and that has generally taken care of the problems related to ASIO driver behaving oddly (or not at all). i'm guessing sometimes the stack of SDK related bits installed gets confused and some of the winapi get used as the wrong version for some software etc... backwards compatibility not withstanding...

very nice work! couple of thoughts: 1) ASIO4ALL - maybe tweak this video or add an appendix video to showing how ASIO4ALL isn't really ASIO so it's fixed in peoples minds... 2) WASAPI is "was-A.P.I." or "W.A.S.A.P.I" not "was-sappy" ? like ASIO is not "ah-see-oh" ?

not all plugins install where the app manager can perform an uninstall, and many don't have an easy uninstall option either (esp if the plugin is installed as part of another software package). using the plugin manager you can see the location of the plugin file and the file name, then simply (using the File Explorer) click on the file, click again (now it's going to let you edit the name), use CRTL+-> or CTRL+END and type ! (so it's now .dll! or vst3!, etc), hit enter to close the edit, and say Yes to window warning you the file may not function... if you have a bulk renamer (i use StEx) you can select all the files you want to drop from the scans, and use the replace .dll with .dll! (or .vst3 with .vst3!). if the plugins have an associated folder, you can do the same thing, just add a ! at the end. then rescan. you should find all the files you appended with the ! are no longer showing up, and yet, if you decide to "restore" the plugin, just remove the ! from the file (and associated directory).

yeah, like HP PC of yore, a number of interfaces need their own ASIO drivers to function. Windows will recognize them in the Device Manager as Audio Interface, etc but only the IO device drivers will work. usually they make a WASAPI driver for general Windows usage, and an ASIO "equiv" for "power users"... not sure how much ASIO is there when they do this. to wit: newer UMC series by Behringer, M-Audio, Focusrite, Avid, etc. many seem to adopt the old "we don't need to use no stinkin' class compliant device drivers" that HP used to do to us, because, ya know? then again, if it's well made device with great features, ok then...

yeah, it would be nice to have the full latency total for a given track and by plugin in a small display or metric popup... hmmmm

in general, you need to know the "maximum" or "largest" latency which is used by the DAW to then set it across the board so everything plays in sync. so if i have only a single track with 5 plugins and none of the other tracks have anything on them, all the tracks will use the latency reported from that one track to set the overall latency so everything plays as expected. e.g. if that one track plugins add 10ms of latency, then all other tracks will be delayed by 10ms so all tracks are playing at the same time. (this is a simplification since internally i'm sure exactly what is happening but the effective results is the same). note that # of buffers used, as well as the sample rate will dictate the overall latency of the chain. CbB has a button "PDC" (plugin delay compensation) which if you turn it off will (should?) remove the compensation added based on what CbB determined based on the latency reported by the plugins etc. this (to me) should identify which track has the most because it will be out of sync. so you'd want to measure the latency of the tracks (or busses) to see which one has the most, and then how much latency that track (or buss) has. one way (manually) would be to use something like https://www.voxengo.com/product/latencydelay/ and pick a track with nothing else on it and adjust the time/samples until it aligns with the longest track. or try out https://www.kvraudio.com/product/vst-plugin-analyser-by-christian-budde some references on internal latency in a DAW: https://www.macprovideo.com/article/recording-and-production/understanding-plug-in-delay-compensation https://help.ableton.com/hc/en-us/articles/360010545559-How-Latency-Works https://www.16sounds.com/blog/zero-latency-plugins/

my son and his friends did a few months of recording using a 1920's washboard (apparently it was too much effort to setup and use my drum kit...) and apparently learned a lot on the way... if you're ok with someone with limited experience in playing a washboard and if he thinks he can fit it into his schedule, i'll have him reach out to you via this board.

as a note about the latency - when using direct monitoring - you'll be playing to the output from the DAW and hearing your drums directly - should all sound normal, however, once in playback, you'll likely find the latency has put the drums out of sync with the other tracks. having an audible click track (use it during recording or not) will make shifting the drum tracks to the proper position easier (rather than simply trying to line up based on the bass or other instrument peaks).

it looks like a muted waveform is there in the circle. read up on how to use the comping tools and you should be able to unmute/unarchive/etc whatever mode that take is in and use it.

in my experience - yes they're saved - you could verify this by setting up your FX rack, saving the file, closing, and re-open it. if the effects have their settings intact, you're all set. as a note, for complex set ups (say TH-3 with multiple amps, cabinets, and effects) i'd save them as a preset in the effect itself to ensure you are in fact saving things, and also you might reuse them as a starting point somewhere else. i'm not sure if saving a track as a "track template" captures all the settings or if they're reset upon importing that template, but again, i tend to err on the side of creating a preset within the effect itself. if the settings are something specific to a project or even a song, i'll save the preset with a name making it clear what it is, and with the project files as well as in the general presets folder.

i don't know of a shared template repository but i suppose if someone asked the moderators if it were possible to set up a persistent thread which people only posted their templates and related info only (a separate thread of people to make comments etc on to keep it down). and i suppose it would probably be nice to have some naming convention agreed to searching for a template could be made easier ?

pretty sure meter changes would be on the measure line, so if you have 3x 4/4 measures, then want to add 2x 3/4 measures, the 3/4 measures would start on the 4th measure: 4:01:000 and return to 4/4 on 6:01:000 not sure you can do it less than full measures...

yeah, that's the challenge. i think a lot of people who don't want to go down the road of mastering the tech in great detail usually learn enough to get a good recording and then take it to someone for mixing and mastering - including vocal corrections etc. one thing always impressed me is watching the John Lennon and Paul McCartney in a studio setting - how they were fascinated by the tech and learned to use it and expand on it in ways people are still trying to emulate to this day. it's somewhat amazing when Paul is doing some demo of his recording and he's just operating all the audio inputs and tape machines or laptop to get his recording... he then has his expert assistance doing the looping, rough mix etc in near real-time as well ?

i only have a single avantone cube for mono listening, and i like it. i use it extensively during mixing. the cool thing is when you final think your mix is sounding good, then you flip to the bigger stereo monitors and - wow! i am thinking of the new NS-10 active clones which have decent reviews.

it's not a bug because it is a manual step, and probably should remain so. making it easier though would be nice to have. i have projects at 44.1, 48, and 96K done at different times. right now, mostly using 48 (mainly because i'm working remotely and only have my remote rig which is limited to 48K and 16-bit) . i would not want my 96/24 files down sampled to 48/24 (files are always 24 bit, even with the 16 bit IO) when i just open it. i'd rather set my IO to 96/24 (assuming it can support, which it does on my home rig). same for a 44.1/24 project - i just set my interface to 44.1/16 (it's limited there but the IO truncation on the output stream is something i live with, the files are still 24 bit). in your case, you bought an interface which only supports 48K. so in that case you'll likely have to change the project sample rates on a project by project basis. presumably you're not editing or planning on editing all 200 projects...

are you rendering the region effect or just trying to "bounce to clip"? you need to render the region fx in order to incorporate the changes. one way (i do this) is to copy (clone) the track, do the region fx and render, then mute/archive the original. this way you can always go back to the original.

hmmm. it is strange. if i use a surround bus in 5.1 (with a stereo track feeding it), and route each channel to a separate physical IO channel (FLCR, Sub, RLR = channel 1-6 on my UMC1820) and it works as expected (i can pan it around a set of 6 speakers, for me to use my dedicated 5.1 cinema system, i have to encode it since it needs a Dolby (or equiv) stream). presumably you could use a stereo track and sends to go directly - you set it up for main out to ch 1-2, c and sub to 3-4, and rear to ch 5-6 - i do this in my UMC1820 and it is different than a surround buss because there is no surround panner - i have to balance it via the main and send levels to position it. i set up a stereo bus like a track - however - sending a track to this buss, there is no surround panner available (same as the track, no surround panner).