CJ Jacobson

4 minutes ago, Murray Webster said:

Mind you, did VVocal reach its apotheosis

I was never a fan of it. It was too clunky for me. Maybe i didn't take enough time to learn it, but that's my take on it

I got ya. then you need to install the full version of sonar with the V-vocal in it what Scook said!!

11 minutes ago, Murray Webster said:

CbB or do I need an old version with VVocal first?

If i remember correctly, V-vocal was not a VST or DX, so you cannot do a custom install of an older version of Sonar and install it in a newer version. It was integrated into the old version.

FYI: Melodyne that is included is far far far better than Rolands V-vocal

But to get all the other plugins (VST and Dx), all you need is the version of sonar you plan to use installed first and then do custom install for the older ones that include the plugins you want

As your input device, you are referring to an audio interface? Yes? If yes, what are your needs and budget? how many inputs do you need? how many mic pre's do you need? how many outputs do you need/ do you need 5pin MIDI? Do you want USB or Thunderbolt? SPDIF connections? ADAT connections?

If you are referring to your Studio Monitors, do you want a sub with them? do you prefer 8" or 6.5" speakers/ whats the size of your room?

CJ

Maybe some sort of RF interference. (electrical interference)

31 minutes ago, gill said:

I need to be able to unlink my audio input and output drivers.  I'm using an AI to record vocals, but I want the sound go go through my computer.  My driver mode is in ASIO, but I can't change my input driver to ASIO4ALL, which is what works for me when I use other DAWs.  Help!

Make 2 different user profiles in your PC. that may work.  One for one sound card and one for the other sound card...

On 4/10/2019 at 10:54 PM, John Nelson said:

Most dynamic processors have a gain reduction meter.  To visualize by means of a waveform... is highly unorthodox, not something mixers typically do.

And in the end the advice is always to use your ears and not your eyes.

+1 , Fabfilter Pro C and L and their EQ's all have before and after waveform views. for me, i tend to just listen to it.

For me, I do not care what it looks like. If it sounds good, its good to go! It can look like anything and it wouldn't bother me

Did someone say Sound-blaster? 

20 minutes ago, Twisted Fingers said:

What can I do to stop the popping and clicking during playback? I'm using the Izotope Production Suite for mixing and I understand that this suite is CPU intensive but my CPU never breaks a sweat.

What kind of processing are you doing? I know with a lot of their RX plugins, there are different real time playback modes. make sure your not at the highest quality, if you are using those types of plugins. I usually select the first or 2nd playback algorithm, but when i export, i select the highest one

On 4/5/2019 at 5:44 PM, tunesmithers said:

am using Cakewalk (by Bandlab) and have SD 3.1.2.  I have noticed that when an instance of SD3 is in my Cakewalk file, the Cakewalk file takes a really long time to open.  Once the project is open in CW, and I open SD3 within, I look at the memory gauge (upper right that shows the kit as it’s loading) and it seems to stall.  For instance, I am loading the NY Avatar kit,the memory is showing 580 as the total RAM, but loaded is only showing 120…  then a minute later it creeps to 150, then a little bit later 168.

Depending on your PC processing speed and RAM IIRC, it can take a minute to load all the samples up.  I remember there being a settings in superior drums for loading samples into ram.

The faster it loads depends on your Ram specs in your PC. Also, if thew samples are not on your audio drive and on an external hard drive, this can take longer

18 minutes ago, Mad Musicologist said:

Only one question @ CJ Jacobson: "You could also have the dropoutsec in your .ini file set to low" - you mean; too low, or should I change it to "low"? Anyway, I did not yet dare to change anything in the .ini file. 

I meant that it could be set to low, but its fixed, so do not worry about it

20 minutes ago, Mad Musicologist said:

Is was the LB-EQ.

Do you mean LP-EQ? The only LB EQ i know is the Neve hardware EQ. Do you have the vst version of that?

Anyway, if that'LB EQ is glitchy, try out the Fab Filter Pro Q3 EQ. I can run over 100 of them in Cakewalk with no problem what so ever and its a great EQ. you can do mid and side and independent left and right channel edits

10 minutes ago, James Argo said:

Well, it's not terrible advice for beginner, in fact "Normalize" is one of the first step to learn how to master.

I do not agree. Normalization is a band-aid and has nothing to do with mastering. Normalization is not used as a tool for learning how to master. 

2 hours ago, girldairy said:

I used to record at home and give it to a mixer. Now I want to export the music(one acoustic track) by myself and found its volume is very low.

I spent half of the day to try all kinds of solutions (Sonar, Audacity......)but doesn't work, the song just cracked if I turn up the volume.  Any advice?

It cracks up because you  are clipping it. You cannot exceed above 0dB in thr digital realm. You really shouldn't to go above -0.03dB to-0.06db to leave room for errors in conversions.

For mastering, you basically just need an EQ --> Compressor --- EQ (sometiems) --- Limiter, those are all you need for mastering. so just learn them inside out and train your ears to your room and wha lah!!!

On 4/6/2019 at 6:52 AM, Mad Musicologist said:

I have a track almost finished, many instruments, an original audio track with the singer, quite some VSt-FX, at the bottom of the mix I have been "collecting" the audio outs to busses, the last before "Master" is "MasterPreview", where I inserted a VSt-FX for room / space, and the LB-EQ to have the mastering section completely in the project.

I see a lot for improvement, but lets start here. Gain staging and signal flow.  The common way to do this:

Tracks go to the Master Bus and Master Bus goes to your Main Outs 1/2.  Sends go to buses and all buses go to the Master bus. 

What effects are you using for those +25dB spikes?

Ok, got that out of the way. 2nd, Drop outs are due to improper driver settings and lack PC performance. so what is your sound card, what settings are they set up to and what are your PC specs. If you have a good PC ,it can be as easy as raising your ASIO buffer to from 64 to 128 or 128 to 256 or 256 to 512 or 512 to 1024.

You could also have the dropoutsec in your .ini file set to low

If you want to use it a a controller to control Fader and bus and send levels, you should get a motorized one. One made for as a controller.  Otherwise, it doesn't remember any of the Fader, Bus, Send and any other settings the project was previously set to and you'll be starting over every time you open and close a project.

Kinda counter-intuitive, yes?

22 hours ago, garylue said:

I've been having an issue lately that I am unable to Split Clips.  The session starts off fine with no problems, then at some point I am unable to Split Clips.  If I restart the program, all returns to normal for a while and then it stops working again.  Not sure what triggers it.  Anyone else having this issue, or have any recommendations?  

Thanks!

I have not noticed a split clip problem.  I key-binded my 'A' to split clips, as my 'S' is key-binded for Save. But its all the same. I think you have a plugin, driver or setting that is freezing your DAW, and its not the split option.

So the next question is, what is in this project and specify audio interface  settings, plugins and anything else pertinent  

Ive finally  got Pyro to work. I just had to run it as an ADMIN and it registered for me. 

Yes there could be something going on in there, if you are having issues with 2 different DAW's. But with Cakewalk, what Noel said is Spot On!!

5 minutes ago, Bruce Searl said:

The volume change from WAV to MP3 doesn't seem like a routing issue if I take the loud WAV file that I exported and convert it with SoundForge. There is no routing options for SoundForge. it's a simple open, convert, save process. 

Its not a simple open and save. Routing involves your Cue Mix also. . I Own and use SoundForge. I get the same gain level when exporting wave files to MP3's.

The only difference is with the lower bitrates, you can get some high end frequencies cut off. so you should use 320 bitrate or 256 bitrate

Your common denominator between your gain issues is your Cue Mix. Your troubles may lie in there. As there is no bug. Its a setting on your part. You just need to find it and correct it.

Is it closing down on every project or just one. This can determine if its a project or the DAW itself

CJ

12 hours ago, Bruce Searl said:

When I do entire mix and everything goes to master on 1/2 out... it will clip and distort. If I mute the other three stereo busses, that do not rout to master 1/2 anyway, it will not clip... so it seems to be a bug. 

I do not see that behavior on my DAW. We have the same DAW. Check all your routing and im sure there is something going on that you are not seeing to one of your other main outs..

I've exported hundreds of audio files in the past week with the latest Bandlab Cakewlak (what ever its called, i just downloaded it last week) and im exporting exactly what i am seeing going out of my master and onto main outs 1/2.

If it was a bug, Everyone would see this, not just you.

Quote

When I open the Wave in Sound forge and convert it to an MP3 it gets softer too... so am I just now realizing that mp3 are not as loud as a wav file?

The volume should be the same. Its routing issue, as you are having volume/Gain issues on 2 different DAW's

2 hours ago, Bruce Searl said:

CJ Jacobson - I playing either form the DAW, through the Focusrite 18i8 1/2 Outputs to my monitors

If you are playing back through the DAW with the same effect chains you exported it out to, you are going to clip. 

2 hours ago, Bruce Searl said:

with Entire Mix selected, has source hardware outputs 1/2, 3/4, 5/6, 7/8 all selected with no way to deselect the extra outputs.

I use entire mix as the source with Bandlab Cakewalk and do not have a clipping issue. As long as your master bus goes to your main outs 1/2, it doesn't matter if 3/4 or 5/6 are also selected. Nothing is going there. It will not make things clip.

4 hours ago, Matthew said:

acks "send fader" is automated, but I still hear the trails of the reverb lingering during that break. Im confused....?

 

So as a test, I brought the "send" fader all the way down so the reverb is not audible. I then listen back to the track with just the volume automation applied and I don't hear any reverb or reverb tails existing on the clip or during the break in the song. So one would assume that by automating the original tracks "Volume" and the original tracks "send fader" would achieve the desired effect im going for, but to no avail.

If anyone can enlighten me on what im missing so I can figure out what im doing wrong. It would be a great help. Thanks all.

A few things that come to mind and to look at:

• Is the send pre or post fader
• You can also automate the bus that the reverb is on with a volume envelope
• I remember a function in Sonar (cakewalk), that you can enable to 'play effect tails' in the preference menu, make sure this is disengaged

Let me know if any of those work,

CJ

12 hours ago, Bruce Searl said:

Since installing and using Cakewalk by Bandlab, when ever I export a project, the resulting 44.1 or 48k wav or mp3 files are clipping and distorted on louder parts or even all the way through a song if the content is pretty loud. Everything sounds and looks fine while mixing and monitoring, no peaking or even near peaking levels. I've never had this problem with Sonar producer or platinum versions. 

Check to make sure your tracks go the the master bus, All buses go to the master bus and that the Master Bus goes to the Main Outs 1/2.

6 hours ago, Bruce Searl said:

porting from the export button rather then file/export and using the Master Mix Preset, and I have the Boost11 Plugin on my master buss, with the output set to -.04db  The reduction that boost11 is saying it's doing is only -1.4db so it's not exactly having to slam the lid down on things.

Do not use presets. Set it yourself. The person who made that preset did not have your song in mind.

Also, ware are you playing it back in?

21 hours ago, kurt said:

Both sound fine  but I am wondering why I need to record at 48K when using  ASIO and why I would use ASIO anyway.

Correct, there are no sound quality differences between ASIO and WDM driver modes. You just use the one that gives you better performance, ass far as your PC and DAW goes.

The only reason i can think of is because ASIO driver that was written was not coded for 44.1kHz maybe. Or maybe you have share drivers with others selected and it needs to be set at 48kHz so other programs can use it as well.

CJ