Asio Puns

Obligatory disclaimer that you shouldn't mess with this if you don't know what you're doing, do it at your own risk yada yada. I don't guarantee that this will work, but I think it could come in handy for a lot of people.

As a short intro, I have had big issues with extremely unstable ASIO performance, where Cubase's Audio Performance meter would randomly spike to overload levels once every five seconds or so, and the audio performance was generally under more stress than it should have been (I even had spikes up to 100% before I had loaded a project sometimes, just because I had a single plugin in my Control Room setup). Monitoring a channel with an amp sim or other somewhat heavy effects would cause lots of sudden clicks and pops which would even end up in the recording and ruin the take (and my mood).

After a lot of troubleshooting, following every guide on the internet, and ultimately experimenting with my own solutions, I finally solved my issues by following the below steps. A caveat here is that this method will not be of any use if your CPU is already struggling to run your projects; in my case my CPU was never over 30-40% even when the audio performance was going haywire, which is why this fix worked for me. It also helps to have as many CPU cores as possible, so make sure to enable HyperThreading or any similar technologies if they are available on your CPU. Feel free to skip the parts within parentheses below if you feel like you know what you are doing, I just wanted to add context so that it doesn't sound like I'm telling you to destroy your computer.

Step 1: Get LatencyMon.

(LatencyMon is a program that can monitor the drivers on your computer, and assess how often they interrupt your CPU. Cubase runs a lot of stuff in real time on the CPU, and all interruptions from other drivers are an issue because the CPU needs to handle them before turning back to processing your audio. There's no way to have no interruptions - all drivers need to access the CPU now and then - but a lot of drivers don't play nice and hog the CPU with tons of long interruptions. Common perpetrators are Nvidia graphics drivers and AMD Raid drivers, for example.)

Simply run LatencyMon (click the green play button to start measuring) and let it run for a couple of minutes (preferrably while you have a Cubase project open just to simulate reality a bit more, just let Cubase sit in the background). When you feel

System Info:

Desktop PC (specs are well above the required/suggested hardware)

Audio Interface Focusrite Scarlet 2i2 (guitar is connected to this using standard guitar cable)

So everything was working mostly fine 2 days ago. I say mostly fine because I was using the acoustic mic method to use my audio interface with the game, which was working, but not as well as I would like. The game was saying I was missing many inputs when I clearly wasn't and/or was not reading the inputs correctly (1 fret off usually). So I looked into ways of addressing this.

On the Customsforge forums, they suggested following the video here:

https://www.youtube.com/watch?v=gIp6IyPjGOY

Which I followed. The only additional thing I did, which may be the problem, even though I tried to undo it; was I tried to use the quick load "mod". The game is installed NVMe drive, so I figured it should have worked...

But since then what happens is the game would boot up, I would get the opening audio, but when the guitar pick would show up in the bottom right all sound cut out completely (including background music playing in an audio player force stopped) and the game would not progress beyond that.

The irony of this is is when looking into getting the audio interface set up I stumbled across a post (may have been here) about this exact issue, specifically audio cutting out during boot, but I can't seem to find it either through a search on here, google, or Customsforge and I'm having no luck finding it in my history either.

RS_ASIO:

>[Config]
>
>EnableWasapiOutputs=1
>
>EnableWasapiInputs=0
>
>EnableAsio=1
>
>[Asio]
>
>; available buffer size modes:
>
>; driver - respect buffer size setting set in the driver
>
>; host - use a buffer size as close as possible as that requested by the host application
>
>; custom - use the buffer size specified in CustomBufferSize field
>
>BufferSizeMode=custom
>
>CustomBufferSize=96
>
>[Asio.Output]
>
>Driver=
>
>BaseChannel=0
>
>AltBaseChannel=
>
>EnableSoftwareEndpointVolumeControl=1
>
>EnableSoftwareMasterVolumeControl=1
>
>SoftwareMasterVolumePercent=100
>
>[Asio.Input.0]
>
>Driver=Focusrite USB ASIO
>
>Channel=0
>
>EnableSoftwareEndpointVolumeControl=1
>

Hello, so I just purchased a focusrite Scarlett 2i2 to record music and interface with my instruments.
As Yousician is always off a bit when using an external mic, I figured I would have the bonus of using the ASIO driver and therefore cut down latency and improve accuracy.
The issue I am having is, it seems there is no way in Yousician to choose a separate input and output, and as I would like my Yousician to output the app sound through my PC which is connected to my Home Audio Theatre device (what I use as speakers for everything else).
This setup works fine when I'm NOT using the ASIO drivers and use Win10Audio, but that defeats the whole purpose of the ASIO setup and the focusrite.

I can get my guitar audio to be recognized by the app just fine when using ASIO focusrite input, but I cannot simultaneously get the app to output sound through my computer, which is becoming frustrating.

I've installed voicemeeter banana hoping there might be some way for me to achieve this, but the app is fairly robust and even after reading the manual I still can't get it to output the Yousician sound to my PC, while using the ASIO input from the focusrite.

Can somebody here with a little experience in such matters please shine a little light on the process and help me figure out how to get this working?
It seems like a huge oversight on the part of Yousician devs that you can't select separate input and ouput in the PC app, unless I'm missing something???

Thank you for whatever help you can give!!!

Hey,

So new proud owner of a katana 50 mkII amp. It sounds so sweet to my ears. Huge fan so far..

It is working with RS_Asio after some installation headaches.

I dont feel it is getting the accuracy of the realtone setup. Missing a lot of notes, etc. Even though I am clearly nailing them visually and audibly. Im thinking its working but not actually. the Log file is jibrish to me though...

Anyone mind looking at my logs? (please)

rs_asio Log https://pastebin.com/euA1tYDU

rs_asio INI https://pastebin.com/9GEXAT1q

rocksmith ini https://pastebin.com/CDEJskbR

Also, 1 huge problem. The output of the whole setup is going through the wrong output. My default output on my pc is set to headphones on the headphone jack. Worked perfectly before with realtone. rs_asio is mixing things up I assume. I cant get it to send through the headphones, no matter what combo in windows 10 i select. (default output, disable the wrong output, etc etc etc)

Much appreciated if anyone has the time to help out here.

thank you

I’m having latency issues with the Waveout audio system on reaper and tried to switch over to ASIO, everytime i do that it sends me an error code saying “There was an error opening the audio hardware… Error getting ASIO sample rate” Any idea on what this is and how to fix it? I’m new to this and just opened up the DAW

Bought the game today, already knowing I could use ASIO instead of Realtone Cable. BUT, using ASIO4ALL v2 Beta 2 and my Presonus Studio 24c, which was listed as functional in the GitHub page, it only wants to use my Virtual 1/2 input, which I never use, instead of my Mic/Inst/Line 1/2 input (managed to spot that it's only using Virtual with the logs when no input sound came, only ouput). How do I work around it?

Edit: prolly should've mentioned I'm on pc

Hey, I have a rather minimal example code using asio, where depending on compiler/operating system/link-order the destructor of boost::io_service blocks indefinitely. Do you have any idea, what could be wrong ?

    // Messenger.h
    class Messenger
    {
    public:
        virtual ~Messenger() = default;

    protected:
        void start();

        void stop();

        using work_guard_type = boost::asio::executor_work_guard<
                boost::asio::io_context::executor_type>;

        std::thread timerThread_;

        // Using optionals to control construction/destruction explicitly.  
        std::optional<boost::asio::io_service> io_;
        std::optional<work_guard_type> work_;
        std::optional<boost::asio::deadline_timer> timer_;
    };
    
    // Messenger.cpp
    void Messenger::start()
    {
        io_.emplace();
        work_.emplace(io_.value().get_executor());
        timer_.emplace(*io_);
        timerThread_ = std::thread([&] { io_->run(); });
    }

    void Messenger::stop()
    {
        io_->stop();
        timerThread_.join();

        work_.reset();
        timer_.reset();
        io_.reset();
    }
    
    // MessengerTest.cpp
    TEST_F(MessengerTest, createXTimes)
    {
        struct Dummy : Messenger
        {
            Dummy() { start(); }

            ~Dummy() { stop(); }
        };
        
        // This blocks indefinitely after 1-4 rounds. 
        algo::repeat(20, [&]() { Dummy d; });
    }

Any ideas ? :)

i heard that asio driver make audio interfaces have low latency, is it possible to get asio drivers on the scarlett solo? if so where can i download them? thanks

Forgive me as I am not the most knowledgeable person when it comes to audio computing...

But for the past few days I have been fiddling around with my guitar & FL Studio, by running my guitar into my FocusRite and selecting the appropriate inputs on the mixer with FL Studio ASIO selected under audio settings. I don't have an adaptor/headphone jack to hear the audio straight from my interface, so this was pretty neat since I could hear my amplified guitar from my computer monitors and mess around with effects. I was even beginning to make some recordings, before I ran into a problem seemingly out of thin air.

I wasn't doing anything out of the ordinary, when FL Studio suddenly crashed, and now FL Studio ASIO doesn't reproduce my guitar anymore, even with the appropriate inputs/outputs selected (& buffer length/sample rate). I tried using ASIO4ALL instead but the input options all say "Not Connected," and yes I have selected my interface on the control panel. I even went into Windows settings, and the appropriate inputs/outputs are all there, I cannot for the life of me tell what the issue is. I could really use some pointers on repairing whatever went wrong, I already nosedived into the image line forums and old reddit threads with no solution to show for it.

Hi all, I am working on building out a UDP server that can receive from multiple clients on a single UDP socket. Please go easy on me, as I am new to networking concepts. For context, I am running on a Windows 10 machine using C++17, and everything is on localhost. My server code looks something like this:

m_socket.async_receive_from(
    asio::buffer(myBufferPtr, myBufferSize),
    m_lastReceivedEndpoint,
    myFunctionToReceiveAgain
);

My client code is similar, using async_send_to instead. Each client and server is given a separateio_context that runs on its own thread, and all of my code works great when I have only a single client connected to the server.

However, when I have multiple clients sending messages to the server at once, my server simply doesn't receive a bunch of my messages. I assumed that this was because the socket was getting clogged, since even though my messages are very small (24 bytes), I am sending them as quickly as my hardware will allow. However, increasing the size of the socket receive buffer by a factor of 100 didn't change the behavior at all. What DID help was implementing a timer on each client to send only at 10ms intervals. This seems to somewhat fix the issue, allowing all of the messages to go through until I have about 5-10 clients. However, any more than that and messages start getting dropped again.

Can anybody enlighten me on what might be causing this issue? Is my approach simple flawed? My understanding is that a UDP server should be able to handle hundreds, if not thousands of connections. Is my throughput simply too high? Is it possible for the contents of my receive buffer to get overwritten between when the buffer is written to and when the completion handler is called? I wouldn't think so, since that would be baffling behavior, but it's my best guess at why simultaneous sends would cause some of my messages to not make it to my completion handler for processing.

I can try to share some code snippets if that would be helpful, but I have a LOT of code building an abstraction layer over ASIO, and I think this is a conceptual problem more than a bug-related one.

Anyway, thanks in advance for the help, y'all are treasures.

So, this is a question I always have whenever I need to configure my DAW.

I have an AudioFuse from Arturia as an audio interface, which I consider a very robust interface for what I need. But I have some synths that I use with digital output. As there is no easy way for me to forward the input signal from my synthesizers to the AudioFuse's ADAT channels on Windows, the best solution I found was to use ASIO4All. With it, I can leave all my instruments connected and access all the channels connected to my computer in my DAW.

However, whenever I configure my DAW, and I put the audio driver as ASIO4All, I keep asking myself if I'm not losing any audio quality by not using the ASIO driver of my audio interface.

I know that ASIO4All will stress my processor a lot more because now both (my processor and my audio interface) will have to manage my audio channels.

But I wanted to know if anyone could tell if there is another problem in using ASIO4All.

This week RS_ASIO and RSMods both got updated releases to include a new feature:

> Disable annoying message when 2 cables are connected in single player mode

If you're an RS_ASIO user, this means you no longer need to change the configuration files when you want to swap between using Single Player or Multi Player. The first players input should be the one as defined with the

> [Asio.Input.0]

block in the RS_ASIO.ini file. Define the second input and under "Asio.Input.1" and you can use Rocksmith without the error message ever appearing. If you find yourself needing to swap the input you're using, you can enter the Multi Player menu in Rocksmith and choose the "Swap Cable Assignment" option.

For those of you that don't use RS_ASIO, but perhaps have 2 Real Tone Cables you want to leave connected to your PC without having to disable one in the Sound Control Panel, you can now open the RSMods.exe, navigate to the Enable / Disable Mods tab, and check the "Bypass 2+ RTC MessageBox".

Windows seems to always assign the most recently connected Real Tone Cable to be the input for Player 1. You can change this in the Multi Player menu with the "Swap Cable Assignment" option, but it does NOT save this between restarts of Rocksmith. If this becomes an issue for you, the easiest way to fix it is within the windows Control Panel and change which of your Real Tone Cables is the "default", or you could just switch the cable you use for single player.

Reminder for RSMods, there are quite a few things possible now, check the tooltips for some information on what each check box does.

Hi, i’m new to music production and am not extremely savvy to the logistics of these softwares yet.

I have a PreSonus Studio 24c audio interface that I’ve literally had for a couple months now and can’t figure out how to use.

tonight I’ve sat down for the past two hours with my electric keyboard, PreSonus, and laptop trying to get ableton to record me playing the piano. it will not.

I’ve downloaded ASIO4ALL, as well as the “universal control” that comes with my audio interface, which I guess installs several different ASIO drivers as well.

I’ve tried recording both with “ASIO4ALL v2” as my audio device and “studio USB ASIO driver” which was one of the ones that came with my interface.

“ASIO4ALL v2” just doesn’t work. I am getting red errors next to my inputs in hardware setup, which supposedly means they are being used by another program, but I have no others running.

“studio USB ASIO driver” seems to work better, in session view when I have an audio track armed I can literally see the input and volume level, but no sound comes out. so I tried recording into the arrangement to see if I’d even get an audio clip, and I did. but I still could not hear a thing when I tried to play it back. I opened youtube and clicked a random video after that, and no sound came from it either.

so then I saved the track which was just one audio clip as a test, disconnected from the interface and restarted my computer, opened ableton, and I could finally hear the audio clip of me playing the piano.

this is obviously a problem as I need to hear what i’m playing or playing along to. in the preferences for the driver I see no option to configure my audio output (I really only use my computer speakers or bluetooth speakers, but I also have HS8’s I can wire it to) except output config, which I already have both options activated in.

please help me. How can I get ableton to actually play sound while using my audio interface with external instruments? I’ve searched already far and wide for an answer but have had no luck. ASIO seems to cause a lot of people problems.

thanks!

I've been getting this error in fl, it play volume from my speakers (through an m-audio 2x2) but when I play through my headphones the volume is always at full blast (doesn't register when I turn down the pc's volume) and the playback line in fl doesn't play.

Any advice?

Hey guys!

I had to buy a UMC22 because my 3.5 jack in the back of the PC produced some squeeking noise. After plugging this UMC22 the sound of the HD599 got worse. It was tuned with realtek audio console for my taste, but with this UMC i cant use equalizer. I tried APO but it doesnt seem to be working.

What to do?

Ive recently switched interfaces from an m-audio mobile-pre to a focusrite scarlett solo. the Solo sounds better, but for some reason is extremely glitchy at the same buffer sizes I used to use on Wave out with the mobile-pre.

I've switched to ASIO and even the reaper says im under 2ms of latency, sound comes out of my speakers MUCH more latent than the reported 23ms I had with Waveout.

input configuration

output configuration

device settings

I cant understand why i cant get the same performance with a newer, better interface. please help :)

Not that there aren't any, but most music playing apps on Linux seem to be built so you have to configure them.

Is there some reason they can't be coded just as a standalone app, so you don't have to configure them specially, or so you don't have to load soundfonts? Or did it just happen that way? For someone coding a standalone music playing with midi app, is there some special difficulty to bear in mind?

My friend gave me his Roland duo-capture mk2 audio interface to use with Rocksmith, but after a day of trying, I couldn't get it to work. After asking for help on reddit and discord, it seems that I've done everything right and the problem is with the roland device.

So I have 2 options now. Either buy a realtone cable or buy another asio that's known to work with Rocksmith. Which one will give better sound quality and latency?

I've got a UA apollo twin TB on my windows machine, and the ASIO driver doesn't seem to be recognized by Via - has anyone successfully made this happen? Not much info out there.

Hi all,

I recently got reaper and I'm new to daws in general, so this may sound nooby. So when I use waveform (I think that's what it's called) reaper works as it should, except the latency it causes makes it unusable. So I switched to ASIO as my audio type and I get no sound at all. I get a signal and no latency from it, put no sound during recording or playback.

After I record something, if I switch back to waveform I can listen to what I recorded in ASIO as well, but it makes it really frustrating going back and forward between waveform and asio.

If any body can help it'd be great, I'm excited to start making songs and this is really annoying me

i heard that asio driver make audio interfaces have low latency, is it possible to get asio drivers on the scarlett solo? if so where can i download them? thanks