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Sudden Latency Onset Issue…

I’ve been creating an issue on subsequent takes; I am calling it the “Sudden Latency Onset Issue” since I can. Basically, if I record an audio track, save the song to disk, then try to replace this audio track, there is an incredible latency between the new audio and the rhythm/beat.
Does that make sense? Has anyone else noticed this or is it just me?
My work around is to basically to not save the song to disk once I have added an audio track until I am completely satisfied with the take.
I am now doing the following steps:
1. Make song 100% except audio track
2. Save to disk
3. Record Audio
4. Review WITHOUT SAVING
5. Accept or reject
a. If rejected, record again
b. If accepted, save to disk,
6. Clean up extraneous files.

Am I nuts, is this what has to happen? I sure hope I’m wrong and seek advice.

Thanks in advance.

PS, I thought I figured it out but was following my steps and recreated the issue so I've edited the steps and I'm trying again....oh well.

Last edited by TomSarge; 05/20/19 08:18 PM.
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What is the program you are using to do this?
What audio driver are you using?
Is the replacement file audio or midi?


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Add nothing that adds nothing to the music.
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You didn't say which application you are doing this in (BIAB? RealBand? PowerTracks? Something else?).

Is your audio interface set to 44.1kbps or 48kbps? All the PGMusic content is 44.1, but if you process it as 48, your audio is going to drift.


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Tell us some specifics about your sound interface, too.

Also, is the latency consistently the same amount of time, or does it drift?

Please don't be put off by our questions; we want to help and we just need to have more to work with.


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Originally Posted By: Guitarhacker
What is the program you are using to do this?

I am using BIAB2019 Pro.

What audio driver are you using?

I am unsure, the default that came with BIAB I suspect.

Is the replacement file audio or midi?


I believe it's audio

Thanks...

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Originally Posted By: jford
You didn't say which application you are doing this in (BIAB? RealBand? PowerTracks? Something else?).


I am using BIAB2019 Pro 64bit


Is your audio interface set to 44.1kbps or 48kbps? All the PGMusic content is 44.1, but if you process it as 48, your audio is going to drift.


I don't know the answer but believe I'm good since it works find unless I save to disk and then try to change it.

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Originally Posted By: Matt Finley
Tell us some specifics about your sound interface, too.

I have a Behringer U-PHORIA UMC22


Also, is the latency consistently the same amount of time, or does it drift?

There is no variation detectable by ear in the amount of time.


Please don't be put off by our questions; we want to help and we just need to have more to work with.




Thanks, it helps to know folks are willing to help me with this....

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There must be some wrong setting. I use the Behringer Uphoria UMC404HD (which is similar to your Uphoria). I have had Windows updates reset the device to 48kbps; I always set it back to 16-bit 44.1kbps and I use 256 samples for recording to avoid MIDI latency (and sometimes increase it for playback of a lot of tracks), but never had the problem of audio being off. This is in both BIAB and RealBand.

You should launch the "Sound" settings in Control Panel and make sure everything is set consistently on the interface on both the playback and record tabs. Also, launch the UMC control panel (which gets installed with the driver) to set you sample rate. That's what I would try first.


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Thanks John,

I always fear that I'm gonna make things worse when I try to do things like this...oh well. But, I have tried.

The settings were different, the speakers were 24 bit, 44100Hz and the microphone was 1 channel, 16 bit, 44100.

I changed them to match, 16 bit, 44100 and I'm hoping the thing doesn't explode on me.

To be honest, the fact that the initial recording seems to work find and only subsequent re-recording causes the problem confuses me but here I go to try and test it. I'll let you know my results soon. If I'm way off base, laugh and let me know so I can try to get back on track.....Into the valley...

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No difference
and I easily recreated the latency issue...

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I've been playing around with recreating the issue and I have found something interesting. I believe there is a temporary file somewhere that isn't getting deleted. Here's why...
Once I create the latency issue, I then created a new song with nothing, no changes, and just counted out the beats...latency issue occurs.
I then closed BIAB and reopened it, created a new song with nothing, no changes, and just counted out the beats....NO LATENCY ISSUE OCCURS.

I am now hunting for temp files in the /bb directory. Oh well.

///// AFTER PLAYING AROUND...

I found three files that are created upon saving/re-recording but manipulation of those files didn't resolve the issue....

bb/TEMPMAIN.bt1
bb/TEMPMAIN.WAV
bb/DATA/RAW_RECORDING.WAV

I deleted/renamed these files and was able to recreate the latency, and when I create a new song without closing BIAB,I get latency. If I close BIAB, new song, no latency. Oh well.
I used to do this sort of stuff on mainframes back in the day, decades ago, but I'm sure it's a temp file issue somewhere. I plan to seek PG help tomorrow....

////////another interesting tidbit...

I created a new song and counted the beats...123422343234 etc. I then recorded ABCDABCD over part of it...The letters have latency but the numbers do not...


Last edited by TomSarge; 05/21/19 05:41 PM.
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Hi
It is my understanding that the latency adjustments in BIAB are just concerned with the screen display staying in sync with what you hear.

The actual latency the time taken to play a note on you keyboard and hear it, depend on the time it takes the computer and the software to make the sound, this round trip time is your heard latency. This can only be affected by playing with buffer times.
The smaller the buffer the less delay but the more stutter from buffer overflow.
Note, even Asio drivers normally have a buffer setting.
Finding the sweet spot for your set up can be a bit pains taking.
Buffer size is often set high to prevent audio stuttering particularly if your pc is doing a lot of stuff in the back ground like internet , anti virus programs, pretty screen effects, so keep all background task to a min and make an exception for BIAB in your AV this should allow you to lower your buffers for less latency.
Just my thoughts
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I don’t think it will fix this particular problem, but every user of digital audio should periodically run the free DPC Latency Checker. Read the web instructions thoroughly. It works fine in Windows 10; it just reads one second too high.


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Which drivers are you using; ASIO, WAS, or MME? BIAB 2019 defaults to wanting to use WAS drivers, but my experience has been it causes problems with the Behringer. I run the latest ASIO drivers from Behringer at 16bit, 44.1kbps,256 samples. I have rock solid recording and playback with no perceptible latency


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Originally Posted By: TomSarge
Originally Posted By: jford
You didn't say which application you are doing this in (BIAB? RealBand? PowerTracks? Something else?).


I am using BIAB2019 Pro 64bit


Is your audio interface set to 44.1kbps or 48kbps? All the PGMusic content is 44.1, but if you process it as 48, your audio is going to drift.


I don't know the answer but believe I'm good since it works find unless I save to disk and then try to change it.


This actually DOES matter. Band in a Box has only one track for you to use to record audio. Real Band allows more. A third party DAW also allows multiple tracks.

Both audio and midi are able to be used in all 3.

When people say they have horrible latency on recorded stuff on playback, it's generally caused by just a few things. The most common is the use of a less than optimal driver, the hardware, or settings.

Some interfaces use wrappers and codecs and proprietary drivers. That can cause serious latency unless set up correctly, and can still be a pin when set up correctly. ASIO is the preferred driver mode if the interface hardware supports it. Generally, if the project is 100% true audio, there are little to no issues in recording or playback. Even MME seems to be able to handle 100% audio projects fairly well. It's where you either have existing midi and synths or introduce soft-synths to the mix with audio, that you encounter issues with latency and the cheaper drivers.

My old laptop would run BB perfectly. No issues, no latency. However, when I tried to run the project in Real Band and added audio tracks, the latency was off the charts bad when I ran it using the default laptop audio driver MME. I could connect my Focusrite interface and use ASIO and the sync was perfect no matter what kind of tracks I used.

Also, variations in the audio track sample rate WILL cause audio tracks to drift of of sync. THis can happen but is not a common issue unless you are getting tracks from outside sources. Everything recorded in your software will have the same settings by default. 44.1 and 16 or 24 bits is standard. 16 or 24 bits are generally 100% compatible and even in the same project will play flawlessly. It's the 44.1khz sampling rate that needs to be the same. I record all my tracks at 24 bits and the software converts them instantly to 16 where needed. 44.1/16 is CD quality and is good for 99.9% of everything you need to do.

First determine the program you are using when you experience this latency. Check the settings to see what driver is in use with the interface you are using. If it supports ASIO, by all means, use it. You will need to experiment to find the problem.

EDIT: I researched this interface and like I suspected, I could not find a single place, including Behringer's website where they stated that this interface used true ASIO drivers. On one, (not the Behringer site) reference was made about ASIO4ALL as a driver but.... A4A is simply a wrapper to use MME and try to fool the hardware into thinking it's running ASIO rather than MME. I would never use any interface that did not support true ASIO. IN other music forums sites, the folks who had the most difficulty with latency were the ones who were often using interfaces that only ran codecs and proprietary drivers and A4A. The solution is to get an interface that runs ASIO. Perhaps you can get the UMC22 working... you can try. And we will help if we can. Settings and check the things mentioned. Good Luck.

Last edited by Guitarhacker; 05/22/19 03:10 AM.

You can find my music at:
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Add nothing that adds nothing to the music.
You can make excuses or you can make progress but not both.

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TomSarge -

You may already know this, but you can go +++ here +++ to get the UMC22 ASIO driver, if you don't already have it.

You still configure the audio device using Windows first; then configure the ASIO driver using the UMC interface you get after installing the driver. Then configure BIAB to use that driver (which will also provide a link to the UMC ASIO control panel).


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TomSarge,

Welcome to the forum and to Band-in-a-Box (BiaB).

You've received lots of guidance and suggestions. I'll add my suggestion as I have not seen anyone else suggest it.

Use the Windows search feature (Ctrl + F) to make sure you don't have an audio file file the same name as your song project. If you do, rename or delete the audio file.

In the example below my song project file is named, "Not_My_Song.SGU" so I am looking for an audio file named "Not_My_Song.wav". If I saved or exported anything as a MP3 or WMA file I would search under "Not_My_Song.MP3" and "Not_My_Song.WMA" also.

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Example screen shot of how to search a computer hard drive for an audio file named, "Not_My_Song".

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I must admit that some of the things folks have suggested are stretching my technical expertise...or lack there of. But, I do know how to search for files as was suggested and...
I am able to recreate the problem before a .WAV file is generated.
Those TEMPMAIN. files are not duplicated.

I'm working with PG support on this as well.

Thanks for your help, I'm gonna try some of the other suggestions now....

Oh well.

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First, let me thank you for your suggestions and research.

I worked on Mainframes back in the day but know very little about PCs. Anyway, if I understand you correctly, you're saying I should not use the ASIO4ALL from Behringer. The post after this suggest I do so I'm not doing anything until I hear from PG support.

Thanks again.

Luck, well, now, so far it's been all bad but I have confidence that this issue is close to being resolved.

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Originally Posted By: TomSarge
......
Anyway, if I understand you correctly, you're saying I should not use the ASIO4ALL from Behringer. The post after this suggest I do so I'm not doing anything until I hear from PG support.

Thanks again.

Luck, well, now, so far it's been all bad but I have confidence that this issue is close to being resolved.




You can try anything you want. You won't break the machine by trying things. I've tried ASIO4ALL... it didn't work, not even close. Just remove the things that don't work. Leaving unworkable things in the computer can cause issues so be sure to get them out. I went in after and totally removed A4A and loaded the native ASIO drivers. Suddenly, the sun came out and the birds started to sing.....

Currently, ASIO is a one trick pony. If you use it on a DAW that DAW will run nicely.... however.... everything else that you load, BB & RB for example, will have to use some other driver...and you will have the issues with those drivers that are inherent in and with them.

If it's any comfort.... it took me several weeks working on getting things set up until I was able to get sound out of the computer, and a few more days working to get things running smoothly. This was back when I was first getting started. SO I've been in your position and I know many others have been too. It was with the assistance of other users that I finally stumbled on the fix. So, keep working on it. There is a solution.


You can find my music at:
www.herbhartley.com
Add nothing that adds nothing to the music.
You can make excuses or you can make progress but not both.

The magic you are looking for is in the work you are avoiding.
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