Glenn Stanton

originally when i saw this a created two tracks with about 30 nodes over the course of 10 or measures of different widths and levels. the relationships stayed the same until i had one or more levels get to the max or min levels, then the proportional thing happens. i'll try some more testing - maybe it's happening when the level shifts are smaller than 1db?

it's linear while the levels are not hitting the top or bottom levels, after which it is proportional, but within range, the relationships stay the same (for me). the example below - originally 0db, -4db, -2db; drag down 0db to -2db, and the -4=-6db, -2=-4db. so for me it's operating like i would expect.

presumably the same - what you set the HPF to is what is allowed to trigger the sidechain. not sure how multiple tracks feeding that will behave as i don't use it that way. but if you test it out, it might be good to share your experience with it.

for a while i was using a separate drive for the active projects and copying over to OneDrive, but i discovered if i turned off onedrive or paused it, it no longer interfered with the project files when working on them. i have gotten a bit lazy after setting it to pause for 8 hours and just let it re-engage after it expires and let it do its synchronization thing... same for the Google Drive which is used for bulk content backup. just re-enable the sync once i'm done installing new content etc and let it work. since it's all mainly working on only updated or new files, it doesn't take as long as the first time...

there is definitely some proportionality going on as one of the sections hits the top or bottom of the range - they start to "compress" or "reduce" until they're all at the top or bottom of the max/min level. i'm not sure it's a bug though.

so, don't confuse the file content with the file representation - a mono recording is "not wide". a stereo recording, if its a single mono source is "not wide", its just "big mono". there are techniques for "widening" mono tracks. and even more for "widening" proper stereo content. sometimes, using mid-side recording (effectively 3 tracks) is excellent for some purposes because you can change the relative width of the content from stereo to mono with excellent mono integrity retention. there is also lot's of discussion on LCR "stereo" (left-center-right) panning and the various degrees within a stereo image.

here's the doc: https://www.cakewalk.com/Documentation?product=SONAR&language=3&help=ProChannel.05.html the HPF allows you to limit the top frequency used to trigger the side chain - let's suppose you have some bleed from a snare on a kick, you might turn the HPF down so the snare is not triggering and only allow the kick signal to trigger.

sure. in dual mono - meaning 2 mono tracks, should sound like a single stereo file when panned center (meaning levels between them are as recorded). and if you have separate pan controls (e.g. protools) on the stereo file (remembering the stereo file is really two separate mono channels in a single file), you have the same concerns about the relationship between the channels as separate mono tracks each having a single pan control. you could easily test this: set up two mics in an XY pattern, as separate channels in your IO unit, record as stereo on one track and record each channel as mono on two tracks. you can do this at the same time. try side by side, same thing except the spill in the mics should be more obvious depending on the distance and source, and then try another. in all cases you should see/hear that both the dual mono and the single stereo are the same. here's a reference for the mic'ing: https://blog.samson.co/blog/2019/08/4-stereo-microphone-techniques-with-audio-samples/ and of course: https://www.dummies.com/?s=+Stereo+Microphone+Techniques

yeah, there is definite physical-transducer related sounds which can only be partially mimic'd by digital. the sheer randomness of overdriven tube resonances, speakers cones, cabinets and proximity of other similar devices (not to mention any quantum entanglement going on between tubes in these devices). that said, i'd suggest that 99.99% of people a) can't tell, b) don't care if it's a proper tube amp or a digital simulation... unless of course you're a rock god and your fans discover they've been hoodwinked because a 1500W 16x 12" Marshall stack was not used... @Lord Tim (you know of what i speak... LOL)

the main difference - dual mono is two files, stereo is typically 1 file. multiple channels (like surround) may be one or more files (depending on if they're encoded or raw really). if you have a single panning control (like the console track) then stereo files will tend to keep their channel relationship. if you have two tracks (dual mono) then your panning could change the relationship (assuming they're not strictly mono - same material in each).

they must not like you much. ? i get at least 6 emails a day on this product... and i only have the NX...

to wit: Marshall amps and Jimi - all dials up, all the way ? oh, and fingers, gotta have the fingers (even Django had 2)...

i just use MuseScore and exchange the MIDI when needed.

ah, wifi speed. has numerous technical impediments - a major one is it's got a radio which is essentially restricted to 54mbs (or 27Mbs) per channel. the way it gets higher speed is using multiple channels... so there needs to be a device on the other end capable of leveraging those multiple channels - hardware of course would be best, but device software can do it as well - just less efficiently. the more devices you have on the network then, the more channels become congested - at the end of the day it is all "serialized" streams... i use 2.4G for "non-essential" and guest wifi, and the 5G connection (not to be confused with the hype about the magic 5G from your phone provider) for important devices (like my desktop/laptop ?) . set the device interface to use fastest bandwidth options, highest power transmit, etc to maximize the connection strength and consume the maximum bandwidth per channel. of course the fastest would be a wired connection running giga speed ethernet through a pair of NIC but that might not be an option unless you have Cat5e (or better) wiring or will run it. forget powerline ethernet...

BandLab should acquire Finale or Sibelius and integrate and be done with it.

i would suggest it be a setting - i would not want the track to force a clip rename or vice-versa unless i enabled it. my track names often share a single clip for arranging and effects, etc so having the track name change to the clip or vice versa would make more work (for me). but there are also times i want to go in and just edit the tracks on a projects and definitely would want the clips to follow suit. so +1

+1

i just had Optimum (north NJ) installed - 300/300 - i was on 100/20 - the difference is radical (but expected by me as having experience building fiber networks). even if i'm at the far end of the house from the wifi router (about 60ft) i'm still doing 200 down and 180 up... i uploaded 2TB (1 each to Google and OneDrive) in about 16 hours... keeping in mind 300mbs = 30Mbs... and yes, at some point the switching circuit where all the fibers connect will have to share. in general a switch consists of say 6 independent sections (often called an "ASIC"), each typically with 12 fibers. if done correctly the sections will be balanced so only a few are truly active at a time - resulting in best overall performance for that switch - but if they stack everyone on a single section before moving on to the next one, it can become congested quickly. always a good idea to ask the installer "hey, can i get my fiber connection on it's own ASIC?" and maybe they can, in which case you'll have some nice speed for a while ? also, ask if the can get the levels to -20db or higher (most times the minimal spec is -24db) (mine is -17db) and this may require a couple of tries on the "fusing" operation (where they connect the raw fiber to the fiber lead going into a TOSLink-type connector box). much more reliable that way...

might be the settings you're using? if you select "all synth audio outputs: mono" (for example) and a "MIDI Source" you might end up with 3 or more tracks...

ok, after reviewing these, you're fired man! ?

i tried to follow the "standard" color scheme for tracks and busses, but, there are too many of them! ? so i compiled them and did it based on % of templates and color schemes... some from pro tools, recording standards, and many contributions from folks regarding how they color their tracks and busses. so let me know what you do with it as i'm always looking out for more normalization of my templates. yeah, the noise is used to listen to a low level (reduced 12db + pink noise) to see what pops out and what's missing. i've only just started using that technique in the last month or so and i've found it handy as a reference. for my monitoring - i use the Wave NX when under the cans.

on the same set of tabs as the media browser - you see the notes icon - you can change the image there. however, i've found that the image doesn't show up in the templates part of the start screen, but once i save a project with that template, it does display it with existing projects part.

doesn't the new track per take in the preference->recording section do that? actually it just creates a single WAV file, but if you then bounce each track take to a clip, it creates a separate WAV file. track 6 (track 1 rec(11) file name) is the original single file of take imported. but if you bounce each to a clips - an individual WAV file is created:

probably need to create a test template (bundle likely) with stock plugins and using various sweeps etc to exercise across a variety of conditions - EQ, compression, reverb, delays, saturation, modulation, levels, pans, tempo changes, automations, groove clips, etc. maybe a bunch of the Pro Channel, SI and other include VI synths, and Sonitus plugins, REW (or other) full spectrum sweeps, noise, and reversed, and say 30 tracks, 5 aux busses, 6 busses, and one set of WAV files for 44.1K@16bit, 48K@24bit, and 96K@32bit clips. loop over say 30 seconds - so there is "settle" time. maybe some marker bursts ever 5 seconds - 1K@-18db. and for extra pain, an academy leader-type video clip that repeats 3x (10 seconds per). that way the complete package for a given test template would be something manageable in size but truly painful on a system ?.

@msmcleod similar experience here. my UMC 1820 (channels 1-12) display in the windows device manager as independent IO but in CbB they're listed as L-R - something like UMC 1820 IN01L, UMC 1820 I02R, etc. the Alesis IO2 - same, and the only time it's right is the generic USB audio codec (and related ASIO4ALL) i use on the Q802USB mixer. on the UMC - i can rename them (and i do like "OUT01, IN02, OUT10, IN03, etc) so it readily matches my patch panel setup. internally, it may be that to keep CbB sane on stereo tracks, so its "hardwired" adjacent pairing 01-02 = stereo track 1, 03-04 = stereo track 2, etc. whereas it would be nice to have an assignment matrix so each stereo track can be assigned whatever IO channels i want - 01, 07 = stereo track 1, 03, 05 = stereo track 2. would likely ease some of the routing for dual mono, and external device connections ?