CJ Jacobson

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.

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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

19 hours ago, David A. White said:

TASCAM US16x08 audio interface because not only does Cakewalk not work with it, but neither does Sound Forge or Band-In-A-Box. They all fail if you use the ASIO driver. They all work with MME or WDM, but who wants to use those?

WDM driver is just fine.

With some sound cards, like the TASCAM,  the ASIO driver mode is a POS.  If WDM gives you better performance, that use it. 

IIRC, TASCAM never had good ASIO drivers for windows

3 hours ago, greg54 said:

 But I don't want to do a clean install every time I need to delete the picture cache.   Or maybe it's something else.  

When I record vocals or guitar, the audio sounds  harsh, and there is a lot of mids, making it sound dark.  

I can assure you, its not your pic cache and it has nothing to do with a clean install either.  A pic cache will not make your sounds sound harsh, and give it extra mid frequency levels. Maybe you have an EQ enabled in the cuemix or maybe you are using a mic like the SM57 that gives you extra mid range. Or maybe sample rate mismatch, but that would make it slower or faster. you did not say that was the case... But its definitely not your pic cache and a clean install.

CJ

1 hour ago, maxsthaven said:

 I do know what you are saying, though, but one will find that doing this creates a worthwhile input difference in a guitar emulation plug-in.  If you’re up to it, give it a try.

You are not understanding the concept of mono and stereo instruments and tracks.

Guitar is mono, so there is no need to change the interleave to mono. Its already set to mono. Guitar is a mono instrument. Even  when you record with 2 mics onto 2 different tracks, you still have 2 mono tracks. 

You do not record a mono signal onto a stereo track and then change the interleave to mono, as you stated above. That's backwards. You record a mono track onto a mono track, like guitar, and the interleave is already set to mono. Hence, no need to change the interleave, even when sending the guitar to a bus with stereo effects. You got it now?

7 hours ago, maxsthaven said:

Signal Flow Quick Tip 1:  set the Interleave to mono when sending a clean guitar recording on to Amplitube or some other guitar emulation; unless, of course, you want it to enter the plug-in in stereo.  It’s easy to forget this.

The guitar is a mono instrument, so the track would be mono already and with the track being mono, the interleave would be mono automatically. You would not have a mono tack on a stereo track. It would enter any plugin mono, even if the effect outputs stereo.

Even if you use 2 mics to record a guitar, each mic would be recorded onto its own mono track.

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3) The sound card (Realtek) is just a chip on the motherboard (i.e. a toy)

If you care about latency, then do not use the on-board sound chip. Get a dedicated audio interface with drivers written for audio production

Yea project template should work for you

21 minutes ago, Mik Urrey said:

P.S. I recreate the project with same samples and FXs, it was rendered very well. But when I add one more sample, I got invalid output vawe.

When I rendered another project with VSTi's and no samples, I got fine quality every time. So I assume now that some of my manipulations with the samples cause these errors.

Try bouncing down or freezing your MIDI to audio before exporting it out of your DAW.  It cannot handle the 64bit processing engine.

FYI: All these problems will go away with a audio interface.

Try making a template with those specific configurations selected...

4 hours ago, Mik Urrey said:

Whichever file format or samplerate or bitrate I choose at the output I hear clicks and pops in the file what I don't hear while I play the track in DAW.

How is your gain stage?

• Do your tracks go to the master bus
• Sends go to buses
• Buses go to the master bus
• Is the master bus the only bus  to go to the main outs?

What are the RMS, LUFS, and peak dB levels on your master bus and main outs read?

Are you 100% positive that your export setting are set up correctly? How are you setting them up?

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Also source samples has this pops and clicks (visible to the naked eye DC offsets) and I can't remove them as DC offsets, using fade tool only.

How to deal with it?

There is a setting in the preference menu to fix DC off set upon audio files, so make sure that is selected

4 hours ago, Mik Urrey said:

Windows 7 x64, built-in audio interface.

You need a dedicated audio interface. The built in sound cards are not able to deal with the demands of producing audio and running a DAW proficiently. This built in card can give you artifacts, as its drivers are lacking and its converters for A/D and D/A are lacking and its internal hardware is lacking. 

2 hours ago, Rick said:

Then something in Windows popped up (running windows 10) saying something about the hard drive.

How much room is left on your hard drive and do you have a dedicated hard drive for only your audio files? That is something you should do, if you do not.

As far as your dropouts, that's just a buffer setting that may be too low for the current work load of your DAW & PC and/or the dropout configuration in your aud.ini file is set to low. I would first raise your buffers, before messing with the dropout setting.

To solve it, just raise your ASIO drivers for your audio interface or slide the WDM slider to the right, if you are using WDM driver mode.

Also make sure you have the latest drivers for your audio interface.. That's it!!

5 hours ago, Royarn said:

What do you think midi buffers should be set at as I have some midi tracks on this so at present.

Since its audio and you are not having any MIDI issues, keep them ware they are set at. The right settings are the Settings that work for you and they are working for you,  yes? 

Are you looping it during playback and is it audio or MIDI you recorded? 

Do you have any Plugin effects on this track? 

This is prob not it, but if its MIDI make sure your tempo map doesn't have any tempo changes and make sure your MIDI buffers are set for optimal response.  You can also try playing with your audio interface ASIO buffers as well

CJ

11 hours ago, jono grant said:

Hi there...

Every time I export some audio out of CBL, the "bounce settings" are set to "mix recall", I keep selecting "Entire Mix" which is what I want, but it always returns to "Mix Recall" and (mix 1 and mix 2 are selected) I'm not even using mix recall.

 

Anybody know why?

Cheers

Jono

I do what BR suggested, but if you want it to default to 'Entire Mix' you should make a project template with it saved like that and it should always be the default.

Pre fader is pre effect and post fader is post effect. Same for send, pre and post sends.

14 minutes ago, micv said:

The issue is often time I don't notice that a section is selected.  To use this new feature I now have do to a 'un-select all' first.  To me there's nothing to be gained here if that's the design.  Am I missing something?  thx

Just click your mouse anywhere on the track view and this will de-select and selections you have made before you export

10 hours ago, micv said:

unchecked the "re-scan existing plug-ins" but it still does a full re-scan. Is there a way to scan just the new plug-in?

You can define the folders it sans in the preference menu.

Just insert he folder location you added the plugin to and delete the rest and it ill only scan hat folder location.

4 minutes ago, MuFever said:

t I mix in -3dB but my audio interface (and propably all the mono devices out there) is 0dB?

No, i'm not saying that. You are confusing pan laws and actual dB levels

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6 minutes ago, MuFever said:

So is there a solution that I mix in -3dB and Master at 0Db so it is calibrated with various audio mono devices out there?

No, no need. You are confusing pan laws and dB levels. They are 2 different things..

1 minute ago, MuFever said:

Yes I did not write it correctly. I am used to -3dB pan leveling that when I turn to 0dB it is like I need to do extra work with track faders.  

I understood you and my response/answer  is still the same:

You mixed the project with the -3 pan laws, so changing the Pan law in a project to something else will change all its levels and make it sound different your ears.. You would need to mix it all over when changing the pan laws.