Old 11-19-2019, 01:14 PM   #1
GreasyBumpkin
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Default Yet another "No sound from ASIO4ALL" thread!

I've been using ASIO4ALL for a while and life was good but now there's no audio incoming on any of the inputs despite all my attempts to troubleshoot.

I followed this guide: http://www.sonicprojects.ch/obx/asio4all.html

still nothing.

The interfaces I am using are: Soundcraft multitrack mixer, Line 6 Helix, Line 6 Stomp, Behringer UMC404,

I use Reaper to track audio and not much else, I also have Ableton, audiomulch and Voicemeter Potato. Is there another way to get my various audio streams into Reaper?

The reason I'm overcomplicating this slightly is that I want to capture my live post-fx mix (which the soundcraft does beautifully), but also want to grab dry tracks from the other interfaces available.

I've considered just setting Reaper to ASIO on the mixer then using Rearoute to capture other ASIO streams from different audio applications. Alternatively I could find some way to open multiple instances of Reaper with different preferences then that would also work. I don't know really, ASIO4ALL was the best solution I had until now.
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Old 11-19-2019, 02:28 PM   #2
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Show us the ASIO4ALL configuration window, the one that looks like this:



What does yours look like? And be sure to press the wrench icon in the lower right corner so you get the full config.
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Old 11-20-2019, 01:12 AM   #3
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This is a problematic idea for a few reasons.

There is no word clock link between the different audio interfaces, so over time they will drift out of sync.

Also you can run into problems where some devices will only work at certain sample rates.

I'm pretty sure the line 6 stomp only works at 48k, so if you are trying to run the aggregated setup at 44.1k it won't work.

Running everything at 48k could fix that, or you could use the always resample 44.1k to on those devices if you want to run everything else at 44.1k.

That option was put into ASIO4ALL to deal with on board AC97 audio devices which were locked to 48k.

Last edited by drumphil; 11-20-2019 at 01:18 AM.
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Old 11-28-2019, 11:50 AM   #4
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HI, here is what my ASIO looks like:
Attached Images
File Type: png HXSTOMP.PNG (23.0 KB, 22 views)
File Type: png MIXER.PNG (25.8 KB, 22 views)
File Type: png ReapAudio.PNG (38.2 KB, 29 views)
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Old 11-28-2019, 11:51 AM   #5
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funny thing though, when I initially removed all the other interfaces bar mixer and stomp, I did in fact get some audio and managed to record it, no crackles or any bad audio, it worked great. Then when I tried introducing other interfaces, it went back to the same problem. Not sure what I did there.
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Old 11-28-2019, 02:51 PM   #6
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Might help if both were set to same sample rate.....you're forcing resample to 48k in ASIO4ll but requesting 44.1 in Reaper.
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Old 11-28-2019, 03:46 PM   #7
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Might help if both were set to same sample rate.....you're forcing resample to 48k in ASIO4ll but requesting 44.1 in Reaper.
So should all the interfaces be synced up as far as the Hz goes?

Does this rule apply to the buffer and samples?
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Old 11-29-2019, 05:55 AM   #8
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Yes, sample rate should be matched. To do this properly the devices should really be synced together but these devices don't have this option. So as drumphill says your devices are not synced, they are running off their own clocks and so you will probably get some record drift between them over time.

It's not really relevant with buffers as the buffer is a function purely of the driver....so if you use ASIO4all in Reaper the buffer you set for ASIO4all is the one all your devices are running at. The reaper request buffer thing (if it works which it won't always) is only changing the driver setting.
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Old 11-29-2019, 07:37 AM   #9
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Originally Posted by Stella645 View Post
Yes, sample rate should be matched. To do this properly the devices should really be synced together but these devices don't have this option. So as drumphill says your devices are not synced, they are running off their own clocks and so you will probably get some record drift between them over time.

It's not really relevant with buffers as the buffer is a function purely of the driver....so if you use ASIO4all in Reaper the buffer you set for ASIO4all is the one all your devices are running at. The reaper request buffer thing (if it works which it won't always) is only changing the driver setting.
I see. OK I will try getting all the hz matched up and report back.

On de-syncing interfaces, what are the consequences of this? If I record tracks will they drift in and out of time with each other?
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Old 11-29-2019, 11:23 AM   #10
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Here is a quote from the ASIO4ALL instruction manual, Section 4:
Quote:
Multi-device-setups require that all the devices involved are running from the same clock source. You can achieve this by daisy-chaining devices via S/PDIF etc. Fortunately, most USB devices will automatically synchronize themselves for as long as the host controllers they are connected to have a common clock source, which is trivially true for the USB host controllers embedded in the south bridge on any mainboard.
So, if your devices are USB-devices they will have a common clock.

For me, this has always worked, and I sometimes use three USB devices simultaneously.
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Old 11-29-2019, 10:18 PM   #11
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Quote:
So, if your devices are USB-devices they will have a common clock.
That is mostly true for the cheapest USB interfaces, but unlikely to be true for most professional interfaces.

Quote:
I see. OK I will try getting all the hz matched up and report back.
For bonus points get all the hz matched to 48K, because some of your devices only work at 48K. If you do that, you don't have to resample the ones that will only do 48K. The other devices don't care. They will run at whatever sample rate, so you may as well run at the rate that matches the devices that can't work at other sample rates.

As for buffer size, I imagine it would be a good idea to run them all at the same buffer size. ASIO4ALL can only present one buffer size to your DAW, and I have no idea how it handles things if each device has a different buffer size.

ASIO4ALL, and the devices aggregated in it should follow the sample rate as set in either the reaper project settings if you have project sample rate ticked, or in audio preferences windows under request sample rate. Sometimes this isn't true, and the device needs the sample rate to be set manually in it's own control panel, but that usually isn't the case.

Using the project settings project sample rate option will override the audio preferences request sample rate option, if you have both ticked at the same time.

So if the project sample rate is set to 44.1K, and the audio preferences request sample rate option is set to 48K, ASIO4ALL will be told to run at 44.1K, and vice versa.

I usually leave the audio preferences request sample rate unticked, and define the sample rate in the project settings.

Last edited by drumphil; 11-29-2019 at 10:36 PM.
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Old 12-07-2019, 05:12 AM   #12
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Ok so the good news is that I managed to get audio coming through !

After 8 different variations of settings I got results with the following:

1. Go into sound management in Windows and set everything to 24bit 48khz as advised

2. Reboot and start up Reaper, go into ASIO4ALL and align all the settings: buffer 512, 32 samples, kernel buffers at 2 (I don't know what these are) and other boxes unticked

3. Request a sample rate of 48k in the audio device preferences

So I saw it mentioned here that I should set sample rate in the project tab but I couldn't find this? Have I misread?

Despite this the UMC404 remained glitchy in a majority of the tests, trying to incorporate the Voicemeter VAIO's didn't go too well either. But I got the line 6, Voicelive and Soundcraft interfaces working just fine, although my PC was slowed down (it's an i7 processor and 32GB RAM so that's a head scratched).

What are anyone's thoughts reading this?
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Old 12-07-2019, 05:55 AM   #13
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Originally Posted by GreasyBumpkin View Post

So I saw it mentioned here that I should set sample rate in the project tab but I couldn't find this? Have I misread?
File/Project Settings but may be irrelevant...leave the box unticked and it has no effect and your general settings will be used.

My thoughts are. "Whaaaat...4 devices with no proper sync on a free asio driver that wraps Windows audio drivers isn't working perfectly...who'd have thought?!"
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Old 12-07-2019, 11:52 AM   #14
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Originally Posted by Stella645 View Post
My thoughts are. "Whaaaat...4 devices with no proper sync on a free asio driver that wraps Windows audio drivers isn't working perfectly...who'd have thought?!"
I'm self taught as far as production goes so this isn't something I would have thought of, it never came up or I missed it when researching.
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Old 12-07-2019, 11:36 PM   #15
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Quote:
So I saw it mentioned here that I should set sample rate in the project tab but I couldn't find this? Have I misread?
File menu --> Project Settings --> Project Sample Rate under the Project Settings tab. Should be at the top left. The first thing you should see when you've clicked File menu --> Project Settings.


Quote:
Despite this the UMC404 remained glitchy in a majority of the tests, trying to incorporate the Voicemeter VAIO's didn't go too well either. But I got the line 6, Voicelive and Soundcraft interfaces working just fine, although my PC was slowed down (it's an i7 processor and 32GB RAM so that's a head scratched).
Using a larger buffer size may reduce the glitches, but ultimately the devices will eventually drift far enough out of sync. It may still work well enough to get a usable recording, but who knows.

Also, why is the added complexity of Voicemeeter needed in this situation?

Actually, thinking about voicemeeter, doesn't voicemeeter banana allow you to aggregate audio devices? Might be worth trying that rather than using ASIO4ALL to aggregate the devices. Perhaps it handles devices devices with no word clock link more gracefully than ASIO4ALL does.

I'd suggest one or the other, but not both at the same time. It's hard enough to get two devices working together without problems, let alone three or four, with a virtual audio cable thrown in for good measure.

EDIT: Had a look at the voicemeeter potato documentation, and I don't think it allows for a enough channels per audio device to do what you want.


Ultimately, the proper answer is to get one audio interface with enough inputs to connect everything.

But I suppose it's worth a try. You may get usable results.


Quote:
although my PC was slowed down
There will be some extra cpu overhead from aggregating the devices, running all their drivers and hardware at the same time. Also, if you're using an existing project that already has 44.1K files in it, reaper will resample them on the fly to the new sample rate of 48K, and this will use extra CPU power.

EDIT: Further thoughts:

Are you using all the channels on your soundcraft mixer? If not, you could use a splitter cable and connect the second output from each device to a spare channel on your mixer, and not put any effects on those channels (or set the recording point to before the effects depending on how the device works).

Exactly what soundcraft mixer do you have?

Last edited by drumphil; 12-07-2019 at 11:55 PM.
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Old 12-08-2019, 07:05 AM   #16
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File menu --> Project Settings --> Project Sample Rate under the Project Settings tab. Should be at the top left. The first thing you should see when you've clicked File menu --> Project Settings.

Using a larger buffer size may reduce the glitches, but ultimately the devices will eventually drift far enough out of sync. It may still work well enough to get a usable recording, but who knows.

Also, why is the added complexity of Voicemeeter needed in this situation?

Actually, thinking about voicemeeter, doesn't voicemeeter banana allow you to aggregate audio devices? Might be worth trying that rather than using ASIO4ALL to aggregate the devices. Perhaps it handles devices devices with no word clock link more gracefully than ASIO4ALL does.

I'd suggest one or the other, but not both at the same time. It's hard enough to get two devices working together without problems, let alone three or four, with a virtual audio cable thrown in for good measure.

EDIT: Had a look at the voicemeeter potato documentation, and I don't think it allows for a enough channels per audio device to do what you want.

Ultimately, the proper answer is to get one audio interface with enough inputs to connect everything.

But I suppose it's worth a try. You may get usable results.

There will be some extra cpu overhead from aggregating the devices, running all their drivers and hardware at the same time. Also, if you're using an existing project that already has 44.1K files in it, reaper will resample them on the fly to the new sample rate of 48K, and this will use extra CPU power.

EDIT: Further thoughts:

Are you using all the channels on your soundcraft mixer? If not, you could use a splitter cable and connect the second output from each device to a spare channel on your mixer, and not put any effects on those channels (or set the recording point to before the effects depending on how the device works).

Exactly what soundcraft mixer do you have?

On Voicemeeter: this was just an app I picked up so I could monitor soft synths without latency while recording (I used to run them on Rearoute, but then noticed an audible delay whenever I recorded and this made it difficult playing alongside live instruments).

I'll head over to the VM forums to see if anyone else has a better idea like you suggested.

It's a Soundcraft Signature MTK12. 3 channels for guitars, 3 channels for vocals, 2 for VST monitoring (VM), 2 for clicks/team chat (things not going to FOH) and finally 2 for the e-drums.

There currently isn't any room for any more inputs but I agree that getting a wet/dry split on the analog mixer would make life simpler. I should take the dimensions of the MTK22 and see if I have the physical space for that on my rig.

There's also ASIO Link Pro which I have yet to try out.
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Old 12-08-2019, 08:24 PM   #17
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Hang on a second!

I just read the manual for your mixer.

The USB audio outputs are taken post gain, but before everything else, so you are already getting the dry recording of each track without effects!

Only channels 13 and 14, which record the master stereo output will have eq and effects from the desk.

Quote:
On Voicemeeter: this was just an app I picked up so I could monitor soft synths without latency while recording (I used to run them on Rearoute, but then noticed an audible delay whenever I recorded and this made it difficult playing alongside live instruments).
Will these synths not run inside reaper?

Another question. Exactly what is the behringer interface being used for? Where does it fit into this picture?

I understand what the line 6 boxes are there for. Guitar goes into them, and then back out into the mixer. But what does the behringer do?

Last edited by drumphil; 12-08-2019 at 08:43 PM.
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Old 12-09-2019, 07:20 AM   #18
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The USB audio outputs are taken post gain, but before everything else, so you are already getting the dry recording of each track without effects!
Yes but no, the mixer is also my live mixer (our whole set up is a like a backline/studio hybrid), it's receiving wet signals from the vocal and guitar processors, with reverbs and modulations, all that stuff. I'm trying to capture the dry signal using the behringer and helix to be re-applied later if needed. Hope that makes sense?

Quote:
Originally Posted by drumphil View Post
Will these synths not run inside reaper?
Yes definitely, but when you click record on the DAW and start playing MIDI live, do you not notice the latency? I certainly do and end up playing out of time.
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Old 12-09-2019, 11:32 AM   #19
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Yes definitely, but when you click record on the DAW and start playing MIDI live, do you not notice the latency?
No. That should be determined by the buffer size and the output latency of the audio device being used at that buffer size.

Are you playing through the same audio interface when you compare reaper to a stand alone synth being run through voicemeeter?

What was the asio buffer size, and which audio interface were you using?

This is getting really hard to keep up with, because I feel like I'm playing 20 questions.

Exactly how did you have things setup when you decided that having the synth in reaper felt slow?

What else changed when you used a standalone synth with voicemeeter?

Honestly, this is hurting my brain. Nobody uses voicemeeter to reduce latency.

The reasons to use voicemeeter include routing audio between different applications or audio interfaces, and having a mixer that gives you some control over those different things. That's it.

If voicemeeter fixes a latency problem, you're doing something else wrong.


Now I'm trying to imagine what else might be going on. Are you trying to record in reaper with a large asio buffer using the soundcraft USB interface, but also play virtual instruments through the behringer at a small buffer size?

And, if you're trying to play your virtual instruments through a different audio interface, why would you need voicemeeter in between the virtual instrument and the audio interface?

Please save me from any more guessing. Exactly how do you run this setup?

Last edited by drumphil; 12-09-2019 at 11:46 AM.
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Old 12-09-2019, 01:58 PM   #20
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OK Drumphil, how about we just go over the Voicemeeter situation in isolation? Because it's not necessarily related to the multiple interfaces thing. Let's just focus on this one issue and hopefully I won't cause you any further confusion.

So when I started getting into home production I only used to use Maschine standalone (it's pretty easy to use, komplete is a good bundle). If you've ever used Maschine you'll know it's not a DAW, but the plugin version leaves a lot to be desired, so running standalone makes for a better workflow.

This lead me to downloading Reaper because it's a DAW that would allow me to record IRL instruments (vocals, guitar etc) from my mixer and also take audio from Maschine standalone via Rearoute. I just wanted to record nothing but live audio from Maschine alongside guitars, vocals and so on, as if playing Maschine was akin to playing your keyboard workstation into the mix.

If any part of that didn't make sense then stop here and let me know or ask me to clarify something.

With this set up I made an aux track in Reaper which took all the Maschine/Rearoute tracks and monitored them through one of my mixer channels in USB RTN mode.

However, I found that whenever I clicked record on Reaper and began playing, there was some latency which lead to the synths being out of time with the IRL instruments.

REMEMBER: the mixer is not just an interface but also an analog mixer, so all those IRL instruments are just being monitored through loudspeakers connected to the mixer, I'm not hearing anything besides Rearoute from Reaper.

So to resolve the latency I used Voicemeeter which monitored VSTs in the hardware output while sending VST tracks into Reaper via the VAIO.

So in summary

Old method:
Maschine > Rearoute > Reaper

New method:
Maschine > 1 & 2

1. Voicemeeter VAIO > Reaper
2. Voicemeeter VAIO > Mixer USB RTN

I don't pretend like this was some kind of genius technique I thought up, it's just a means to an end I figured out while observing what other people do on YouTube and forums. The goal was merely to monitor my VSTs in a way that made it comfortable to play alongside other musicians in the room, and not to have to use that awful Maschine plugin.

Hope that part at least makes sense now

Last edited by GreasyBumpkin; 12-09-2019 at 02:53 PM.
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