Asio buffer size reddit. that unit sounds super coloured.
Asio buffer size reddit. Sample rate also affects latency.
Asio buffer size reddit Now that I'm using an audio interface (Audient id4 mkII), I switched to an ASIO driver using my audient id4 as the source. not possible, but…. The buffer will determine how many samples of delay you hear between the DAW's timeline clock and your speakers. ASIO already has very low latency, so realistically your CPU speed limiting the buffer size won't be an Hi everyone, I just bought a new audio interface, but I can’t change the buffer size in Fl, even if I press the ASIO control panel it remains locked at 96smp. Menu "Shut Down Voicemeeter" and relaunch Voicemeeter Banana. Hello all, Question for any advanced asio4all users - for the last few months, I have an issue with asio4all where static and crackling slowly accumulate over time. I am using an M-Audio Keystation 49, Saffire 6 USB, and Live 8 Lite. Jul 30, 2015 · And, in the ASIO buffer settings for the TUSB Asio driver, should I set it to minimum latency and the largest buffer, or minimum latency and moderate-size buffer, or what, for the smoothest experience with, say, watching movies? What about the smoothest experience for music in players other than Foobar (PotPlayer, let's say). Sometimes it can go to 32 as well. When I wanted to change it from 512 to 256 it wouldn't, instead cakewalk would crash. However as I mentioned the minute a PT session opens up, Pro Tools changes the buffer size on the Focusrite Control Panel. EnableWasapiInputs=1. Small buffer sizes result in the least latency. cubase and studio one, both have a second buffer you can leverage, so all the playback tracks use a much larger buffer, and the live monitoring tracks run at your sound card buffer size. The higher the buffer size the higher the latency. 3. Yes a larger buffer uses more memory, but it's a trivial amount at any buffer size. To fix it I need to go to Preferences > Device > Asio Configuration > Buffer Settings and set my Preferred ASIO buffer size. This is still happening. Alternately, you could send things through the built-in Windows audio system, which as far as I'm aware automatically adjusts the buffer size and has a On both my Ryzen 2700X 8-core desktop and my i7-4720HQ 4-core laptop I was able to run the same buffer settings (changing from an UM2 to AXE I/O allowed me to lower the settings). With that said, we're talking difference of maybe 1. Recording at 44. Suddenly this week it's showing a red exclamation mark and glitching during playback with ableton. Remember to make sure you have a large enough buffer size set (generally this is set in your audio interface's settings, but it depends on your gear). Reply reply That's a sample rate not a buffer size. This isn't really how it works. 67 millisecond delay -- completely unnoticeable, even if you were to do very time sensitive things, like monitoring I immediately turned down the sample size and am now happily enjoying gaming and listening alike with latency under 2ms. It can't even record through WASAPI. Having installed FlexASIO, I can now set buffer size down to 48 samples, record mic input, record with loopback. If you max out your CPU often, go a bit higher on the buffer settings. 8ms roundtrip at 128. Each time I open Reaper, and occasionally when I record, my Buffer seems to get reset. The smaller the buffer size, the less the latency. I have a godly computer, so I can't seem to find the problem. 5ms. In Pro Tools, however, it immediately changes the selected buffer size to a smaller buffer size when I try to change it. I'm using a MIDI keyboard to play a soundfont and getting lots of pops and cracks. I regularly run my RME HDSPe AIO at 32/64 samples and usually raise to 256 towards finishing a huge project. 1s to say or so it feels like) Hey i got the 4th gen scarlett solo and it works with this config. I recommend getting understanding on how these numbers matter. It's even better to work on the native Windows ASIO drivers at this point than through my Focusrite 2i2o. Sounds like a buffer size issue. If you installed correctly the drivers from behringer's page, the device should be listed as ASIO something and not as "windows audio", that's probably why you are experiencing too much latency. It adds latency. ASIO Buffer Size What does this effect? I've heard to leave it at 512, but what would having a higher number do? I've heard to leave it at 512, but what would having a higher number do? Or lower? I have a inter i9 9900k and 64Gb of Ram + an M-Audio 192|6 interface (ASIO). If you want to use other buffer-sizes, DirectX/MME Drivers can set any buffer-size but their performance will be terrible. The buffer size on cakewalk won't change. So I was wondering what the best value was to set them at. Data is written to the output/playback buffer in a quick burst and then it flows-out to the DAC at a smooth-constant rate. I am using a scarlet solo ii, and it I didn’t have this issue with my old Mac. I've literally tried almost everything and i couldnt find any sort of asio thing or panel that has a higher buffer size than 2048, if there is some driver i could use that has a higher block size then that would be crazy Using asio4all, get crackling and static that slowly builds over time, dependent on buffer size. It feels timing/synchro related more than performance related. In Asio, start reducing your buffer size to as little as possible before you start hearing distortion. My PC is 64gb RAM and i9 10-core processor, which should definitely be able to handle a much lower buffer size. g. Since KA6 ASIO only goes to 1024 if that's still not enough I switch to FL Studio ASIO and use 2048/triple buffer for 123ms total. So at 128 samples in your buffer, at 48KHz in your DAW(or "48,000 samples/sec" if you want to think of it that way), your buffer is 128 out of 48,000 samples. With A4A, there was an inbuilt GUI to change buffer size. Sound travels 343 m/s in the air at a normal room temp range. Please help! I use Bitwig, it kinda supports WASAPI, but it uses pre-win 10 APIs, which don't support setting custom buffer size in the shared mode and loopback recording neither. Sep 5, 2013 · ASIO4ALL is the absolute worst ASIO driver I've used for latency. Indeed, by augmenting the frequency rate as well, you increase the data flow even more. Is there another ASIO driver that has a lower default sample rate/lets you change it and have multple programs outputting audio? Thanks Firstly, you need to use ASIO drivers. KA6 actually goes down to something stupid like 32 samples but I never go that low since it's only an extra 1 or 2 ms and I'll just have to turn it up in a minute anyway. I've been having latency issues while recording vocals. As for how your setup works. I just helped someone configuring live+obs+voicemeeter and with his focusrite-something he have buffer underun at 44Khz, 512 buffer. Low buffer = less delay = less distracting if you're monitoring in headphones while you play. It won't stop popping and clicking. The latency is horrible when playing instruments via my controller and I could never get down to the sample buffer size that I have with ASIO installed. Also, you shouldn't really need to change buffer size once you've found a suitable one. I’ve been using a 2048 buffer size by default without giving it a second thought, I do all my mixing, recording and exporting without changing it. Some drivers multiply the set latency or add an additional buffer. My latency went down from 192ms to 10ms. Additionally, when I come back to the Focusrite settings, I typically find that the buffer size has reverted to a default setting (192), rather than the setting I left it in the last time I changed it. [Config] EnableWasapi=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=48 [Asio. In, and your Monitor to Auto (Figure 4). e. I've upped my buffer size to 1024 and it fixes the problem. In Ableton, I can easily change the buffer size by selecting the buffer size in the control panel's 'Buffer Settings'. Products, practices, and stories about the profession or hobby of recording, editing, and… Open the ASIO panel and in the ASIO's settings select your normal devices (Realtek microphone and speakers, or whatever. SO I dive into the Focusrite settings and adjust the buffer to literally anything else, and audio comes back on. “Buffer” size is doing literally what the word buffer means. what should i set the usb streaming mode and asio buffer size to get the best sound quality in movies and games ? right now i have usb streaming mode on minimum latency and asio buffer size on 32768 samples. So, I have DT990 Pro 250 ohms connected to an Audient iD4 MKII and under the settings it says I can change the ASIO buffer size and sample rate of my headphones. 384k is massive overkill. It sounds like you're trying to change buffer size on the fly while streaming/recording? I don't know of many ASIO devices (or any, actually) that are capable of that. I'm trying to record at a lower buffer and mix at a higher one. it’s called ASIO guard in cubase, can’t remember what it’s called in S1. Extra info: [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 My default settings on the Focusrite control panel of the pc is always 48khz Sample Rate and 512 Samples for Buffer Size and its always been that way. You may experience drop outs, clicks or pops due to buffer underruns. NOTE: When used the underrun counter is bypassed and buffer underruns may be more audible. The option allows some audio devices to reach lower latencies. It’s doesn’t lag. I can *usually* also have it a 64 samples but sometimes the cracks and pops show up due to the extra overhead of ASIO link pro so I sometimes have to change it to 128 samples. ASIO driver will help but you really need a dedicated audio interface (like your Focusrite 2i2) to get to acceptable latencies. I don't know if selecting multiple outputs or multiple inputs works. Higher buffer size increases latency and decreases CPU load. The lower the buffer size, the lower the latency, but the higher the load on your CPU. Have a Google around and you’ll find some detailed info of you need it. Reducing buffer does reduce latency - because you are reducing the pre-creates samples. bs 64 = very slow / saturated sound (rocksmith guy in intro takes 2 seconds to say rocksmith) bs 128 = almost good but tinny sound with crackling (best settiong) bs 256 = sound is sped up (a lot, rocksmith takes 0. I can hardly monitor. When using ASIO link pro to stream audio over zoom, OBS etc. 0. But this also counts for recorded audio going back into your session, so 5. toml file that lets you/3rd party GUI tweak this, but it's not there for me. It just increases the buffer size a bit to help prevent audio issues/artifacts if the processor gets bogged down. 64bytes buffer at 192khz will give you the most intense data flow with the lowest latency. Also rocksmith recommed 128 buffer size , i would set audio engine to 2-3 and bit rate to 16 ( if no crackling after new settings, maybe change back to 24bit for better app compatibility and dynamic sound. Always go in multiples of 64 as a general rule. [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=42 [Asio. I also have a Behringer interface, set to 64 Samples and 96 kHz, which *in theory* would be 64/96000 = ~0,67 ms each for input/output latency. If I switch to ASIO4ALL It opens up and lets me adjust it but not for my mbox. So I go to my audio settings (I have CoreASIO set as Input / Output), click on "Show ASIO panel", but I can't change the buffer size because it's all greyed out and seems to be stuck on 16smp (1ms). Even if my buffer is 2048. FA doesn't. 1 or 48 and keep your buffer at 512, ensure the asio driver for your focusrite is selected. Asio buffer size Discussion This in the picture is the control panel of Dac smsl su1, I would like to know what value to set the "preferred Asio buffer size" on Keep your rate at 44. The size of the ASIO buffer and the associated latency cannot be changed. 0-1. The lower the buffer size, the more cpu power you use, just be aware of that. Sample rate also affects latency. That's like 4ms round trip latency at 44. Hi everyone, I installed amplitube on my gaming pc, which could easily handle low buffer sizes for live play. I have 1 input and 1 output device selected). There's supposedly a . Bigger buffer sizes can help smooth out glitches in the audio stream, but results in more latency. ASIO 4 ALL doesn't count as it isnt a true ASIO driver and was designed as a last resort driver for compatibility for integrated chipsets. Pro Tools 12 can't even handle 512 Buffer Size and always hits CPU limit ASIO is not only overrated it's woefully outdated and simply should not be part of any modern audio system. Reaper's stock gate/EQ/comp are very lightweight and safe to use while tracking. View community ranking In the Top 1% of largest communities on Reddit. The higher the sample rate, the lower the latency, but again higher sample rates use more cpu power. Get the Reddit app Scan this QR code to download the app now The audio interface is Scralett 18i8 2nd gen and I have no problem setting the ASIO buffer size to 16 here is my RS_ASIO Config: [Config] EnableWasapiOutputs=1. I have a Scarlett 2i2, Rode Procaster and a cloudlifter and I've noticed that if I had my buffer size over 64 on Focusrite Control, people would hear a lot of distortion coming from my end. I'm afraid there is no other answer to this that buffer-sizes other than the one listed in Live are not supported by ASIO drivers in Live. if this doesn't happen it means you gotta download your audio interface drivers. [Config] EnableWasapiOutputs=0 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 To me it seems like a firmware issue with the interface. I would try out a buffer size of like 1024 and bump up if you notice issues. The audio would breakdown without the proce 571K subscribers in the audioengineering community. But the reality is that the driver adds another buffer, and the interface also has a hardware buffer inside for sending/receiving the USB data. Works great so far. 9ms. If you are recording and monitoring the recording from the saw you want as low of a buffer size you can get without causing your computer to crash or stutter audio playback. I use the Focusrite ASIO driver normally for my interface. The tradeoff for the "safety" is that increasing the buffer increases the latency as well. this will show up the driver window, where you can change the buffer size. [Config] EnableWasapiOutputs=0 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=driver CustomBufferSize= I have have persistent crackling and pops from my interface, despite using an audio buffer size of 512. FocusRite has its own ASIO drivers optimized for the FocusRite system(s). Hey everyone, I need some help. Set the buffer size in the ASIO settings and the Max Latency in the ASIO Bridge settings. I've also tried to select 144 block size in Reaper instead of selecting the buffering size in the ASIO control panel, that i can access in the DAW, but nothing I have a fairly cheap roland quadcapture and it's set at 48khz with a buffer size of 256 and it run fine. 26 votes, 41 comments. The best FL Studio resource on the internet! I’ve tried updating my audio drivers, increasing the buffer size, reducing the sample rate, freezing tracks etc… My gear consistent of : Windows 10 pc with 32 Gb of RAM, Ryzen 7, Ableton 11 standard, Focusrite 2i2. Yes it does. Is there another ASIO driver that has a lower default sample rate/lets you change it and have multple programs outputting audio? Thanks I use the highest buffer size (2048) when using my audio interface because I use some very heavy plugins, and with a lower buffer size the audio playback in Ableton will start crackling. Is there something I can do to use the 256 buffer size without crackling? I'm using a PIONEER FLX-6 Controller btw May 4, 2021 · The best I can do for ASIO buffer size is 64 samples when just using the focusrite driver. Ofcourse that all depends on the way the rest of the computer works. On another note, definitely drop your sample rate to like 44khz or something. Mix in buffer switch - Output audio is mixed in ASIO's 'buffer switch'. 1983, though maybe I never had my buffer set to 1024. Triple buffering is Hey everyone, Just wondering if anyone had experience using XLR mics and/or Scarlett 2i2 with Discord. In Ableton select Asio as your driver type and Asio4all v2 as your output device (Figure 2) Set your Audio from to Ext. I'm using the default CoreAudio, but I saw a windows machine that is 10% less powerful than mine, running at 256 without sweating, but the driver was ASIO. The best setting for Buffer size is generally the lowest you can get away with before you get stuttering/audio dropouts. If it falls behind for a millisecond, but can catch up in the next, the buffer will prevent dropout. Either I don't understand how this works, or there's some setting I must be missing, but the buffer size is set at a constant 2048 samples. Hey, everyone! So I recently got the Volt 2 and ran into a pretty serious issue. 1k with a 128 buffer? 128/44,100 = 2. Sample rate and buffer size trade cpu and memory. Latencies are pretty low, but not exceptionally so. If you don't have any audio interfaces with specifically made drivers for it, you can install ASIO4ALL which achieves a similar result, but you may find Turn on test tone, up CPU Usage simulator to 80%, play around with the buffer size until it starts being audible on the test tone, then set the buffer just above audible level. Shortening the sample rate can produce clicks and pops because of the increased cpu demand. There are some things you can do, also, adjust the buffer size for recording. Audio system: ASIO ASIO Driver: Focus-rite USB ASIO All In's and Out's enabled Sample rate: 48k Block Size (currently): 1024. If your computer is even modestly powerful then you're just causing unnecessary latency by bringing it up. 1kHz including the USB safety buffer. I'm frustrated since nothing has seemingly changed in my setup. The buffer size really only matters in audio production. But oh dear that unit sounds super coloured. To find your optimum buffer size turn up the CPU simulator to 80%, activate the test tone and increase the buffer size until the tone can be heard without any artifacts. It's zippier than "Primary Sound Driver" in FL, though. And a computer upgrade will get you what you want (I’m guessing it’s the one bottlenecking) the audient evo 4 is a nice interface specially for the price. i use the sennheiser hd 660s for movies and games mainly. If the buffer did change while you're tracking to avoid under-run, that means the latency would change too, which might affect your playing, and I wonder whether a (when using rs_asio, the other tweakable parameter is the interface's buffer size, how many individual samples are collected before handed over to the program, 128 should be doable without much problem even on a laptop - complements the latencybuffer settings) [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=driver . Render some parts down to audio so it's not having to calculate all the effects in real time. However, I don’t have the option to choose any buffer size lower than 256ms, which is way too high to play live with. , with its own Realtek dialog box for ASIO latency/buffer setting). Unless you are getting skipping, then the buffer is already adequate. When I used to use an MME driver, I could modify the buffer size within Ableton. Even with my buffer size and 512, or even 1024, I still get occasional crackles. All legit. The buffer I use is usually 512 samples or 1028, but even then the problem sometimes still occurs. So really, what you want is the smallest buffer possible that still allows for glitch-free Jan 21, 2022 · So, for instance, 96 KHz with a 4096 sample buffer size should give you: 4096 / 96000 = 0. I've had the Behringer UMC22, with the asio4All drivers Behringer recommends I wasn't able to go below 256 samples buffer size. I set up the ASIO4ALL driver (witch is the only one aviable for this particular interface), but i can't change the buffer size in the Ableton preferences menu, just in the ASIO settings. Generally you can't through ASIO, I think that's the highest buffer size ASIO drivers ever have. No joke - I think they're the best in the business! And their ASIO driver is not only simple to install, it really does make the difference with their equipment! Posted by u/giwhS - 1 vote and no comments Rule of thumb: the higher the ASIO Guard setting, the less strain on your CPU, but with added potential for real-time issues (MIDI fiddlin' etc). You made it sound like you were just adjusting the ASIO buffer/block size in your ASIO driver settings? Which is 'preferences > audio > device > ASIO configuration' Whereas the 'preferences > audio > buffering' page I'm referring to contains settings regarding how Reaper utilises your CPU and disk. The ins and outs of this are far too much for here. You'll notice buffer in youtube video - where youtube preloads video data, for you to view video continuously. I'm using an ASUS Xonar Essence STX sound card right now and have had it set to a buffer size of 512 samples for a long time. 11. I never had the problem before updating to Focusrite Control v. I'm closing this conversation as there is nothing to be added to this. Find, download, and install it for the very best in ASIO interface compatibility. Thanks! Jun 19, 2022 · A buffer is also a delay. ASIO Buffer Size (Windows) I can’t change my buffer size, my pc is brand new and I have tried The mic runs with ASIO driver and has an option where you can put 144 buffering size which if i select it and then monitor with Reaper in real time, my audio gets completely distorted. You should now hear yourself through the mic, through the headphone jack on the Yeti. Buffer size is measured in samples, which is the number of samples you're using from your samplerate. 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 I have no experience with SSL2+ don't know how good the driver is and what is your target buffer size? I'd imagine it should run stable 64-128 samples buffer, given you are using i9 cpu. Output] We would like to show you a description here but the site won’t allow us. If you get tripped up on the lowest buffer, nudge it up one level (ie from 16 to 32) Some plugins are too heavy to run at a low buffer size. Output When it's installed, it becomes a visible ASIO option for sound apps and DAWs (I. 13) Open your Voicemeeter options, check "System tray (Run at start up)", keep Buffering ASIO at Default (It will use ASIO4ALL buffer size), sample rate at 44100Hz and change "Virtual ASIO Type:" to Int32LSB. 67 milliseconds of delay. All the best, This also happens when using the FL Studio ASIO driver. Jul 9, 2021 · I would just leave it at default. He has the link inside. I can set the buffer size to 128 samples without dropouts. If I use the FL Studio ASIO on 256 it gives me 6ms but the playing feels really weird. There's a constant static noise that is very loud coming through the outputs. Hey Guys, I'm pretty new to ableton and recording and bought myself the Behringer UMC22 Interface. Always worked for me. EDIT: For any future people here's what helped me solve my issue/my line of thinking: Playing around with the ASIO buffer size slows down / speeds up the ingame sound. Nor any FA file that lets me change this. usb streaming mode and asio buffer size. It's driving me mad because I can't work (mix) like this. One or more DPC routines that belong to a driver running in your system appear to be executing for too long. This happens if the buffer is too small or if something hogs the system for a few-milliseconds too long. At 32 samples it starts producing droputs regularly. 412K subscribers in the FL_Studio community. I get crackling even with one or two VSTs that aren't even heavy. The fixed buffer size used by the Steinberg built-in ASIO Driver should allow a sufficiently low latency. [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 All the buffer size does is give the DAW a little bit of wiggle room. You need a faster system (which can either mean better hardware or better optimized) or a lighter workload to run smaller buffer sizes. The reason to adjust the buffer size is to switch between these states. Only until it dies again. Say I select a buffer size of 1024 samples, this results in the buffer size being changed to 512 samples. It was designed to solve a problem back in the days of Windows 2000 and Windows 98, and the way it shoehorns into the system is bad for overall system stability. One problem may be related to power management, disable CPU throttling settings in Control Panel and BIOS setup. I have it set to 512 samples- any higher or lower and it freaks out and just lags pops, and clicks even more! With A4A, there was an inbuilt GUI to change buffer size. ASIO4All presents an ASIO-compatible interface to apps like Reaper, and translates it to the underlying WDM/WASAPI interface to the device. So with more buffer you’ll get bigger latency. On the playback engine of Pro Tools, the buffer size is constantly 512. If you can increase your buffer size by 8x, your memory cost will increase by 8 times, your latency will increase by 8x, but your cpu will drop. My buffer size is set to 256 SMP but will not open up box to change when I click on it. Google 'realtek asio' and look for a result from baumannmusic. It’s allowing the system more time to do calculations. It's actually inside an older Dell realtek driver package. I originally installed Asio4all as it reduced all of the latency issues, but all of a sudden it started giving me this weird issue about how the sound card was being used and I could not change buffer size until I shut off the program using it. Changing the sample rate won't reduce latency, you should look at the buffer size. I usually have it set to 64 or 128. If Reaper doesn't jive with the control panel setting (it usually autodetects the buffer size), you may need to restart it. 1024 samples at 96 KHz, which should give you a 10. Triple buffer - Can reduce audible underruns when close to 100% CPU load with some ASIO drivers. Shorter buffers require more cpu 🤷♂️ but the data flows faster. you gotta click on "hardware setup". Your computer should be fine with e. If the buffer isn't re-filled in time you get buffer underflow and a glitch/gap in the audio. 04267 seconds, or 42. Plug it in, go into the audio settings in ezdrummer3 set the rate to whatevever the highest is and then reduce the sample size to the lowest you can without distortion (~300 for me) You seem to understand how buffer length correlates with latency correctly, so all I would advise from here is you can adjust your sample rate lower or higher than what I said depending on whether you want less latency or fewer underruns. I recommend you reduce the buffer size. At 64 sampels I get occasional droputs. How low you can go depends on your CPU among other things. Just try the lowest setting available and see if you get dropouts - if you do, increase the buffer. If you set buffer to 64 samples, it will wait til 64 samples are created and then only will start sending to DAC. My cpu should be able to easily have the buffer at less than 128. Tl;dr: Turn down sample size. bvlzpndndjsbwjpbzficpihwjocaiwljsmfnwfoqgjvaclot