Glenn Stanton

@Noel Borthwick just did an experiment -- there is definitely a +3db increase when exporting the clips. pan law set to 0db centered created a sine @ 440hz -15db RMS (12-db peak) for 15 seconds .WAV import into CW (drag and drop) -- check levels = -12db export clip = -8.99db (this is the 3db increate) imported the exported clip - as expect now -9db pan law set to -3db sine/cosine imported clip - now -12db now export clip with pan law set to -3db same - +3db added pan law setting has no impact on the export or import of the actual clip. source file Cursor position (Time) 00:00:00.000 Sample value at cursor (dB) -Inf. Minimum sample position (Time) 00:00:00.019 Minimum sample value (dB) -12.000 Maximum sample position (Time) 00:00:00.030 Maximum sample value (dB) -12.000 RMS level (dB) -15.010 Average value (dB) -Inf. Zero crossings (Hz) 440.00 exported clip content: pan law set to 0db center Cursor position (Time) 00:00:00.000 Sample value at cursor (dB) -Inf. Minimum sample position (Time) 00:00:00.019 Minimum sample value (dB) -8.990 Maximum sample position (Time) 00:00:00.030 Maximum sample value (dB) -8.990 RMS level (dB) -12.000 Average value (dB) -Inf. Zero crossings (Hz) 440.00 pan law set to -3db center sine/cosine Cursor position (Time) 00:00:00.000 Sample value at cursor (dB) -Inf. Minimum sample position (Time) 00:00:00.019 Minimum sample value (dB) -8.990 Maximum sample position (Time) 00:00:00.030 Maximum sample value (dB) -8.990 RMS level (dB) -12.000 Average value (dB) -Inf. Zero crossings (Hz) 440.00

in the export dialog you can pick the export type as clips, i used {clipname} in the example - TRACK001 has CLIP001 and TRACK002 had CLIP002, and then exported. clips came out as named and types. note that the clip name is also in the underlying track lane so it may be possible to select a variety of clips from the lanes to export... would presume that if i exported the stereo clip as mono i'd still see the +3db -- so maybe this doesn't solve the problem -- not sure the pan law settings would be applied but conceptually you could set it to -3db center and see.

if you have Melodyne installed you could try that to see how it works. i've done some instrument pitch changes on guitar and vocals. with care the change is transparent. I have the studio edition so there are features in that version not necessarily available in less expensive ones. i also use it for MIDI extraction so i can replace instruments or vocals with instruments.

I generally agree - but my Q802USB has no ASIO driver, and the WASAPI does a decent job, but i've found the ASIO4ALL goes lower latency than the WASAPI and so i use it only to record. when using my main IO (UMC1820) then i use the native ASIO driver. ? the biggest win (for me) was deleting all the bogus drivers installed by applications (some of which were simply rebranded ASIO4ALL) and only having the native WASAPI, my Behringer ASIO, and the ASIO4ALL (rarely used).

you could set the stereo buss to mono then drop -3db. the merging of stereo into a single mono is typically +3db (just the nature of db math...). i recently "re-discovered" this as i was doing some mono checks and accidentally exported as mono. then importing into Ozone the waveform was higher than expected (doubled effectively) and was purely mono... lol.

the vocals are actually from my original demo and i chopped them up and melodyned the crap out of them to get them to sound reasonably close. i was going to only use them as a reference and re-record them, but after listening to them, i thought, hey - these are kind of quirky in a good way so i'll leave them for now. mainly the only other effect was long delays on the "clean water" section and then reverb as part of the overall room. on the "viking march" at the end, the bom-boms - i used melodyne to drop them low and auto-panned them. i had done that in the demo and it was freakish so i had to keep the concept... update: using my Synth V vocals with background singers as well. in this recording - the instruments are all computer generated with some tweaking, except the solo. i used HookPad, Scaler, patterns, etc to create the tracks, and in HookPad where i used some instruments, i substituted some of those for other sounds. so when i started writing this song (or parts), it as 2016. then a close friend passed in 2018, and as i worked on it, i took some of the things that were part of our life experiences and shaped lyrics around that - our wiccan / earth magic beliefs, drunkeness / etc, blues music, etc and it seemed to stream together. i put the demo together (mainly played live on acoustic guitar and double bass) in 2019 but didn't really record it "proper" until late last year. --- yes i know Quetzalcoatl is mispronounced ?

since i haven't spent much time on the internals of audio programming on Windows, this last post sort of sent me down the rabbit hole: https://learn.microsoft.com/en-us/windows-hardware/drivers/audio/low-latency-audio & https://learn.microsoft.com/en-us/windows-hardware/drivers/audio/ ? lol. one note: if the ASIO driver depends on the lowly WDM then it explains why i can use the ASIO4ALL driver (which uses WDM) for very low latency captures (<5ms) vs ~5ms on WASAPI Exclusive or Shared (my Behringer Q802USB doesn't have a native ASIO like my UMC1820 does which has ~2ms), and after cleaning up all the other drivers (extraneous from Intel, Real, Magix, FL, Avid, Ableton etc -- seems everyone has a driver) off my laptop system, and installing my Behringer UMC and ASIO4ALL drivers fresh, i have no issues with audio on my laptop regardless of mode (other than i have to switch off the ASIO4ALL once i record something and go back to my normal WASAPI or UMC ASIO settings - takes just a minute of time to close the project, switch, reopen).

according to UAD, WDM and ASIO drivers are different - so it would be important to select the correct one - it sounds like you select the ASIO driver in CW but use the WDM version for Windows. maybe set the ASIO on for both? Using Windows WDM System Audio with UAD Hardware – Universal Audio Support Home (uaudio.com) https://help.uaudio.com/hc/en-us/articles/360051064291-Using-Windows-WDM-System-Audio-with-UAD-Hardware as far as UAD and MMCSS -- probably a good question for their tech support.

updated - additional lyrics and new singers including backup ? 70 BPM ~ G Major / E Minor ? ------------------------------------------------------- breathe in the air reach for the stars heal everyone and dream with your heart then the road to valhalla lies clear there's nothing left to fear oh oh i'll let bacchus decide how we spend tonight Quetzalcoatl will call the tune as we dance in love around the moon Hera says that all is fine cause i am yours and you are mine Gaia dances for all to see don't mess with her ecology cause i did hear a wiseman say please don't let it slip away clean waters still matter clean waters still matter help those in need reach for the stars feed everyone and dream with your heart then the road to valhalla lies clear there's nothing left to fear oh oh i'll let bacchus decide how we spend tonight Quetzalcoatl will call the tune as we dance in love around the moon Hera says that all is fine cause i am yours and you are mine Gaia dances for all to see don't mess with her ecology cause i did hear a wiseman say please don't let it slip away peace and love still matters peace and love still matters breathe in the air

when i do use this technique - i use one of my -12db pink noise files imported into an audio track, output to master buss, loop it (make sure it has proper beat count), and set it to -13db - which turns out to be -14db overall. then adjust each track - initial set to 0, and each buss set to -inf. one at a time raise until you just hear it, then mute it and the next, repeat until all done, then unmute them and make some final tweaks before turning off the noise. most times i find the mix to be pretty close and now ready for some tweak via automation. one thing - after using noise, take a walk for some beverage or bio break before re-listening as the noise means you should "re-calibrate" your ears...

maybe document the steps you use to recreate the issue - intermittent issues are tough to solve ? and then see if it's something the support folks could recreate on their end and thus be able to fix it.

Mixing To A Pink Noise Reference (soundonsound.com) https://www.soundonsound.com/techniques/mixing-pink-noise-reference

maybe describe the problem so people who don't want to download the video can understand what you believe the issue is? maybe it's something with take lanes?

i use a PRINT stereo buss which is fed from the MASTER via a pre-fader send (level=0) and its output is set to None. then in the export i select the PRINT buss instead of MASTER. I can set the PRINT level independent of the MASTER. The level for PRINT could be via the send or the volume on PRINT. Note that MASTER has a light 2-buss compressor w/ 2-3db swing on the meter for some light glue.

unfortunately i don't think here is a generally automated way to get reliable results. one thing i use is the Melodyne level balance macro to increase quiet sounds and reduce louder sounds: https://helpcenter.celemony.com/M5/doc/melodyneStudio5/en/M5tour_MacroLeveling?env=standAlone i believe it's available in all versions except the essentials.

yes, so either prep them using a tool to adjust gain to a consistent level, then import. or in the DAW, use clip gain to adjust the entire clip or just parts of it. or if destructive gain setting is ok, then use the process->apply effects->gain and set it up or down.

bottom selection of the amplitude tool. on my own materials, i'll simple select all, then use this tool to drop all sibilance 2db or so (starting point is around -10% then up or down slightly). just enough to start the process without being obvious. on the help page they have some ideas on split tracks to adjust the "balance" of the sibilance -- https://helpcenter.celemony.com/M5/doc/melodyneStudio5/en/M5tour_ToolAmplitude_Fade_Sibilance?env=standAlone

definitely - there are really three levels measured in the LUFS/LU - the "integrated" (you need to be at or below the spec at the end of the total sound recording). "short term" (which has a time window 3000ms), and "momentary" (very short measurement window) ? then you have the loudness range (usually in LU) which gives you some idea of the dynamic range. one thing i've observed in mastering forums and online tutorials, blogs etc is that many mastering engineers (i am not one, nor do i play one anywhere) will do a bunch of their magic to get the integrated value to be where it needs to be (mainly critical for broadcast because in some countries there can be monetary penalties) and to accommodate the behaviors of streaming services automatic leveling "feature", and then leverage higher short term and momentary levels to get the overall level to be perceived as "loud" without violating the integrated requirement. definitely different than simply looking at peaks and RMS. the Izotope Insights meter is very nice, the TC metering is also very nice, and the free YouLEAN meter is decent. some free ones i've played with vary in their accuracy.

select clip(s), select "process -> apply effects -> normalize" (or gain depending) and set the parameters and hit ok. each clip selected will have the normalization performed. depending on your goal, clip gain can be very useful for leveling and overall changing level as well

4. allow use of arpeggiator effects (arp, strum, etc) on select chords ?

i like to use RX on client vocals to clean up as part of prep and do some light de-essing. on my own, i will use RX and then clip gain and edits, then use Melodyne to reduce some of the sibilance, and then one or two de-essers in serial to get sss and cha and tsk which may be different sets of frequencies. each stage is generally small and may be on the VOX buss if i have some doubling, or as Tim noted, on the BVOX buss for background vocals. otherwise, it's mainly the primary vocal track.

i'm not seeing this behavior in my setup. maybe set the "ExceptionHandlingSeverity" = 7 in the initialization section of the Preferences and capture a crash dump.

perhaps this may be helpful -- if you have several monitoring devices setup (e.g. your main speakers, headphones, a BT speaker, etc) it is a good idea to set them so they have the same output levels - in my setup (currently my laptop on my work desk) i have a USB mixer (headphones), a USB Spark Mini, and a BT speaker. first I run a -12db RMS filtered (pink) noise signal from CW (with the outputs on the track, buss, and h/w all at 0 - so my peak level is ~-12db on the meters) and set all my device output levels to 83db -- this may require each Windows sound setting for a given speaker to be adjusted -- my mixer is set to 16, my BT speaker is set to 80, my Spark is set to 66 and each position on their volume dials are marked (permanent marker or nail polish drop). note: if you are using a subwoofer it's a bit more complex because you want to adjust the cross-over(s), delays, and so on to blend with the speakers - phase and transition from the speaker to the sub. then i set the output devices to output 75db (my typical mix level) and mark their dials (or for the BT one i know i press down volume 3x). if i want to go lower i can just move the volume control or pass it through my -12db master channel in CW template. then go through each audio playback software - Media Player, Dopamine, WinAMP, Groove, VLC, etc and set their volume to match 83db when playing the noise file. lastly, adjust my online media hosting players to also play at the 83db level. now i can just adjust the output devices to either 83db or 75db easily and know each player or source has been "calibrated". a decent sound level meter is useful (a long time ago i bought an old Radio Shack digital one for $30 on ebay) and place it approx where you have your head. for my normal full setup i put it on a stand but at my work desk i just hold it there. it's close enough. the whole process only takes about 20-30 minutes, but you'll be much less frustrated when switching across listening devices and players. one update: your decision on source level vs monitoring level can vary based on the system you're using K, LUFS, etc so create (or obtain) the source level file you need (e.g. -18db, -23db, and noise type) and set your full-scale monitoring as defined in the system. in my regular setup, i have a monitor controller as well as a number of presets for the sound levels documented so if i'm doing video / theater / surround vs audio stream, i can switch to proper monitoring. here's a handy reference to some of the different specs - https://www.tcelectronic.com/loudness.html ?

depending on the failure - i've found many times that the power supply and/or power regulator in those types of units are the problem. so, for like 6-7 failed drives (several brands) devices now, i remove the disk drive itself and use a USB adapter unit (which also supplies power for older drives), then copy all of into my cloud storage (2 different services for redundancy). label the disk and put it into a static-free bag. it's something similar to this: https://smile.amazon.com/Converter-External-Universal-Function-Software/dp/B00EHDTRJ6/ which i purchased about 4 years ago. only one drive was not recoverable using this method as the disk drive controller was shot, so i sent to a data recovery place which was able to directly access the drive (it was a small HDD and i didn't have any other like it or i would have swapped the controller board...) that was $150 for the service and they returned the content on a new disk drive (200Mb).

yeah this would be nice so instead of having a temp track with a bunch of sends to busses and etc to be able to insert a set of busses. what would be really really (really) nice would be the track colors (in a track template) could be preserved with the buss assigned color... what i do now is take a cwp with only the busses and rename it as cwx but that is a pain to do if i want it sync'd with my mix template.