Mark Morgon-Shaw

30 years and I did not know number 5 ?

Freeze only works at track level not clip level. It's designed to alleviate the CPU load when you have a lot of plugins and VSTi's running Freezing at clip level wouldn't help with CPU load as the plugins would still be running if only part of the track was frozen You can bounce a clip to audio and it will automatically create a new track for you containing the bounce.

It's very easy to bounce a clip to audio in Cakewalk, you can bind it to a shortcut as well like so. https://gifyu.com/image/e99s

I'm not talking about tracks exported for other musicicans to import into another DAW in order to record more tracks with, obviously you'd want those to be spot on. I'm talking about final exports of finished tracks. Also you wouldn't get further phase issues if the same offset was applied to everything. However a lot of folks in home studios don't consider phase issues when recording or mixing either. Years ago I had an argument with a guy on a collab site because his snare was 180 degrees out of phase and he couldn't hear it in his mix. Eventually after he'd called me everything under the sun he came back and said he'd gotten someone round to check his studio and there was a fault in the internal wiring of his compressor which flipped the phase....he thanked me in the end. ?

No - It would just need adding to the export. Here is an excerpt from the publishers FAQ guidlines for deliverables. ----snip ---- ARE THERE OTHER TECHNICAL SPECIFICATIONS? Yes, there are. As there will be no further mastering, all tracks should (1) have a maximum peak-level of -0.3 dB - NOT normalized to 0 dB to avoid “intersample overs” that cause clipping in D/A converters and in mp3 conversion. (2) All tracks should have an Loudness-Level of 12 LU/-11 LUFS or an RMS-level of -12 dB (equals a dynamic range of 12 dB) – of course classical music will have more quiet parts than e.g. house or rock music, so try to adjust the loudest parts of each track to -12 LU/dB RMS. All tracks (3) should have a balanced level of high, mid and low frequencies. (4) They should have 0,25 seconds of silence at the beginning (some D/A converters in CD players need some msecs to work properly) and 0,25 seconds of silence at the end. (5) All tracks should be checked for clippings, dropouts, “clicks”, “pops”, cut reverbs at the end and other errors. (6) All Tracks should be checked for mono compatibility / phase problems. ---snip---- So imagine that you have to export a stem for every track cue they accept which is usually 5 - 10 exports, then the different alt mixes which is 3 or 4 normally, you're deliver at least 10 exports per track so using a stereo editor to as 250ms to the file after the fact is just too time consuming ( based on my own 100 tracks per year output which is fairly typical for the industry ) Currently I get around it starting the track at measure 2 and then exporting a fraction of the silence at the end of bar 1 ( usually a 16th of a bar will do it ) So the option to add it automatcially as part of the preset would mean I could start my tracks at bar 1 which is easier because you get used ot the math of tracks being in 4 or 8 bar sections - plus it could be exactly 250ms, not some time that quite close but is different dpending on the track tempo.

Yes this is a good idea

It works here, Here is a recent track where I used it on the guitars - most of the clips need to go down slightly in the mix apart from one near the end that had to come up It sounds like you are doing the right thing..split the cips..Ctrl-drag the red line up and down. Not sure why you are are not hearing the level changes Unless you are adding FX that are bringing the level back up ? Compression, limiting etc ? Try bypassing all the FX and listening again (blue FX button on tool bar ) and see if the changes are more obvious without the downstream FX ?

Yes, I've listened to HiFi setups that cost in excess of £100k and it's a completely different level of experience. But the demo recordings they use are not by people making music in their spare rooms. They're generally top class musicians recorded in world class faciilites with high end equipement playing great music. Not Jon from Basingstoke in his shed making sub par Prog drivel on his laptop so he can get 7 likes on Facebook. So it's all relative. Songwriting - Performance - Production - MIxing - Mastering all trump recording media and delivery format. I can get more enjoyment from a great song that was recorded 70 years ago in mono , over an AM radio than virtually every home produced self released track I've ever heard. And I've heard a lot. I agree that N'th degree improvements can wring every last ounce of detail if you have the ears and the equipment top hear it , but if it's not top notch material to begin with then nobody to whom it might matter will be listening.

It's not that subtle to me either, but I spend hours every day listening to audio critically. It's what I do. The rest of the family wouldn't tell the difference, they are casual listeners who consume music on their phones, laptops or smart speakers or possibly ICE system. I don't even think my teenagers have ever heard a lossless format. They are typical listeners and they will never ever hear the difference in a mix. However if you give my wife a photo which looks perfectly good to me she will point out if the focus is soft, pick up on chromatic abberations, converging verticals and blown highlights. As experts we focus on tiny details which are insignificant to the uncaring masses. Hence..moot.

I never said you can't prove it exists , what I said was it's moot because 99.9% of the listeners can't hear it - hell ...most of them can't tell 128k MP3 from a CD.

Yeah, I prefer the midi learn thing but it loses it's bindings if you clsoe the project..which seems a bit mad

All these high sample rate , high oversampling options are pretty moot in a real world scenario. Most people can't out-mix 48k 24bit...and if they can most listeners can't hear it or lack the equipment to resolve such n'th degree differences.

I've always seemed to struggle with ACT, I just don't bother now

One of the publishers I work with ask for a short silence of at least 250ms at the start of each track so this would be handy, especially if it could be added to the Preset

Yup

I think what's happening here is the cursor ( which is invisible at this point ) is hitting some sort barrier and stopping It's not that the automation line won't go any higher because you can release it , grab it again from a bit further down and eventually drag it all the way to the top. Glad to hear you've had the same issue with the lop-sided nodes on the Ctrl-drag Another one I've encountered ( and posted about previously to no avail ) is that sometimes you can't drag a track into a folder from above. It misses the Folder altogether and jumps to below the Folder. Then you have to release it and drag it into the Folder from beneath.

I've had my Stream Deck for quite a while now but I set it up with my own commands, I didn't know there was a company selling dedicated plugins - it looks quite clever. When I first set mine up I added things like all the transport controls etc but for me I didn't find it that useful to control things I could easily access with the mouse and keyboard in the end. It was just easier to hit the spacebar. However I use it all the time for things that are a little more hidden away that I use quite ofte.. Eg. I have one button to open my Audio interface driver so I can adjust latency A macro that closes the Browser and the Inspector and sets Cakewalk to full screen A couple of Cal routines set to their own button like Humanize A few buttons to use in the Track View , one opens the Scale Velocity dialog , another 2 transpose the selected clip up or down an Octave which is about 6 key presses in one button so it is quite a time saver. I also set one up to bounce the selected clip to audio which I find pretty handy. The cool thing about these devices is how flexible they are and they're generic unlike a dedicated DAW controller so I can use it for other Apps as well Hope you enjoy using it.

Seems to work okay for me in a current project However...... I have been having different issue so I wonder if it's related In theory I should be able to select those two upper most automation nodes and Ctrl-Drag that line between the two points up and down. But it always seems to " let go " of the right hand node , and I end up with this instead It always used to work fine but at some point it's gone a bit strange. I thought it might be me but if there are other known issues then perhaps it's related.

It's a perfectly valid attitude, very few musicians can wrap their head around the concept. Even fewer can actually come up with the goods consistently. But relating back to the OP , when there's deadlines involved and you have to pump out a lot of music in a variety of genres you don't get the luxury of waiting for your muse to grab you. Creativity is like a muscle and the more you work it the easier it becomes, a lot of it is to do with overcoming resistance and procrastination. I don't think it's especially age related , maybe the end result is different at different stages of life but I make more music now than I ever did in my teens / 20s / 30s

It would make much more sense to me if the two were independant

Another +1 for Izotope RX I have Waves de-clicker which can sometimes work but RX is a whole different level

I don't think it is ASIO, if you watch this video ( FA-06 / FA-07 / FA-08 all seem to share the same driver ) he installs it on SPLAT and it looks as though it's just a class compliant generic driver.

I use the baby of the range the ID4....can't say I have had any issues like you have mentioned..not sure if they use a common driver ? Have made a screengrab. This is the one I use and I can go as low as 32 samples buffer if I don't want to run hardly any plugins - 128 is where I normally start but I use lots of plugins so I generally end up at 1024 near the end of the mix process. This the resplendence Latencymon result after playing a track with about 25 tracks and 35 plugins Mine is a desktop PC with a Ryzen 3900x - hopefully that gives you something you can compare against to see if anything is different.

So basically you have :- Vamp 3 - Amp Sim which it says comes with a UCA222 USB audio interface for stereo I/O and direct (zero-latency) monitoring - this ships with ASIO4ALL because Behringer use it instead of writing their own drivers on their budget products R8 - Mulitrack Recorder which also works as an ASIO interface Roland FA-07 - Workstation keyboard with Multitrack recorder and USB connection ( not ASIO AFAIK ) I think the first thing you should do is install the latest ASIO driver for the Zoom and get it working and use it as your main interface and mixer etc. The Roland is mainly for midi I assume so that can just stay as the controller keyboard over USB. The Vamp 3 is using the cheap UCA222 to get it's audio into the PC and it's only got ASIO4ALL so I would ditch that little interface and hook the Vamp up to the R8 and go in that way.