24 bit 44.1khz is all you need, buffer size is essentially the amount of latency (time you allow for your computer to process the audio) and increasing it increases that latency but decreases cost on your CPU. It may not display this or other websites correctly. Note: Larger buffer sizes will also increase the audio latency. It's really unbearable! Increasing your buffer volume helps because it ensures data is accessible for processing when the CPU needs it. Even the slightest delay in sending just one out of the millions of samples in an audio recording would cause a dropout. The CPU, RAM, connection type, interface in use, and simultaneous channels can all affect what buffer size is needed. You can usually raise the buffer size up to 256 samples without detecting much latency in the signal. I hope you found this post on what buffer size is good for recording, helpful! on_and_off Well-written driver code manages the systems resources more efficiently, allowing the buffer size to be kept low without imposing a heavy load on the computers central processing unit. Any system that employs pitch-to-MIDI detection, such as a MIDI guitar, is also prone to noticeable latency on low notes, as it needs to see an entire waveform cycle in order to detect the pitch. However, its common usage to refer to this code collectively as the driver.) You mean "buffer size", not sample rate. That combo should 'stick'. And in any case, we may want to choose a different sample rate for other reasonsmost audio for video, for example, needs to be at 48kHz. I can get to 32 samples on an i9900k with an RME UFX+, but I generally hang out on 64. For the lowest monitoring latency, set it as small as you can get it without incurring dropouts, glitches or clicks. Most DAWs offer six buffer size options: 32, 64, 128, 256, 512, and 1024. Focusrite, Apogee, and Universal Audio are three companies who make great quality interfaces, but there are plenty more for you to check out! It's easy! I'm using the Focusrite USB audio driver as the audio driver. This type of arrangement has a lot to recommend it when youre recording bands live. In this video, I want to show you how Buffer size and Latency can affect your recording in your DAW. Purchase Soundkits and more - http://bit.ly/2QcRX2A . There's no absolute answer to it as a lot of factors are involved. Also - one of these days I may finally pull the trigger on an RME PCI card. Started 1 hour ago Next, increase the buffer size to 1024. Nevertheless, many players complain that even this amount of latency is detectable; and there are situations where much smaller amounts of latency are audible. But this line of thinking opens up another discussion: do computers behave as magnetic tapes, in which there was a difference in sound quality among different brands? In general, it is therefore good practice not to introduce any plug-ins that cause delays until the mixing stage is reached, although not all recording programs make it easy to find out whether a particular plug-in adds extra latency. 25th March 2014 #21. . Set the buffer size to a lower amount to reduce the amount of latency for more accurate monitoring. Any technical advantage that, say, Thunderbolt has over USB is only meaningful in practice if the manufacturer can exploit it in their driver code. Approximate latency for common buffer sizes and sample rates. How much latency is acceptable? Go with 96000/32 in the Focusrite setting. WAV vs MP3 vs AAC vs AIFF. Higher sample rates can have advantages for professional music and audio production work, but many professionals work at 44.1 kHz. By amazinjoe555 July 2, 2020 in Audio . For instance, if we are monitoring input signals through an analogue console and the level is too hot for the audio interface its attached to, the recorded signal will be audibly and unpleasantly distorted even though what the artist hears in his or her headphones sounds fine. The driver and related software are critically important to achieving good low-latency performance. If youre not monitoring exactly whats being recorded, you leave open the potential for things to go wrong in ways that can only be discovered when its too late. Is this issue even related to buffer size. Press question mark to learn the rest of the keyboard shortcuts. Your email address will not be published. Rumman When my projects get heavy, I always make sure to turn that on. When I'm not in the studio, I bring my Babyface with me and leave the converter behind since I don't usually do surround nor need lots of IOs when travelling. A less well-known fact is that recording software itself adds a small amount of latency. I usually use 32 samples, or sometimes 64 samples (for high-res, high-track-count situations) when . Rammdustries LLC is compensated for referring traffic and business to these companies. When latency creeps above a few milliseconds, it quickly becomes audible and can badly affect performers. 2. Always use a value expressed in powers of two; 32, 64, 128, 256, 512, 1024. One reason why Apple computers are popular for music recording is that Mac OS includes a system called Core Audio, which has been designed with this sort of need in mind. If youre using the same plug-in on multiple tracks (e.g., a reverb on vocals or drums), then create a bus, route all the tracks there, and add the plug-in. I cant believe how low I can go with buffers and how small the latency is. Reddit and its partners use cookies and similar technologies to provide you with a better experience. This is common practice in large studios, where an analogue mixing console is often used as a front end for a computer-based recording system. Well, doing the sums says that with 256 as the buffer size, you'll end up with 5.8ms latency. (Technically, the driver is only a small part of the code that enables recording software to communicate with recording hardware. The key to achieving unnoticeably low levels of latency in the studio is to choose the right audio interface: not only one that sounds good and has the features you need, but which will be capable of running at low buffer sizes without overwhelming your studio computer. However, if it doesnt and you want to figure out the amount of latency at the current buffer size and sample rate, then divide the buffer size by the sample rate as mentioned above. The downside to lowering the buffer size is that it puts more pressure on your computers processors and forces them to work harder. As for buffer size, I tend to use the largest I can get away with give what I'm working on. When we use a MIDI device to trigger audio in a software instrument, that audio only has to pass through the output buffer, so experiences only half of the usual system latency. Why can't this conversion be extended to include 88.2k, 96k, 176.4k, and 192k? Steinbergs ASIO Direct Monitoring is probably the most widely supported of these, but it is far from being a universal standard, and other solutions require the user to choose both hardware and software from the same manufacturer. The only way to ensure that those sounds emerge promptly when we press a key or twang a string is to make the system latency as low as possible. When it comes to latency, you cant always believe what your audio interface is telling your recording software. One other thing to remember is the Direct Monitoring switch on the 2i2. 2 Mic/Line/Instrument Preamps. They let us apply EQ, compression and effects to more channels than would be possible in any analogue studio. If the performance improves, you can try a lower setting. You'll know only when you try :|. Therefore you may notice audio dropouts at lower buffer sizes, depending on the overall CPU load of the set. and high buffer size when mixing/mastering. Best Buffer Size For Mixing & Recording [Buffer Size Explained] Orpheus Audio Academy 2.1K subscribers Subscribe 127 Share 6.8K views 1 year ago ++ SONG-FINISHING CHECKLIST ++ (Finish more. Would I be safe at 64 for example? What sounds too low? We set down the latency to 89 samples buffer size (producing a global latency of 13.9 ms which is much bigger than expected for this buffer size). How Does It Work? Posted in Power Supplies, By 64 buffers in so incredibly low - why are you wanting / needing it to be lower? Linus Media Group is not associated with these services. This means that if any problem occurs further along in the recording chain, we wont hear it until its too late. What Is a Digital Audio Workstation (DAW)? A device called an analogue-to-digital converter then measures or samples this fluctuating voltage at regular intervals44,100 times per second, in the case of CD-quality audioand reports these measurements as a series of numbers. However, not always the highest number means the best option. Right now my settings are 48K sample rate and 128 buffer. A quick representation of the same waveform being sampled at different settings. So, if you have a computer that only has 8GB of RAM, then your computer may struggle recording at 88.2kHz sample rate and a buffer size of 64 samples. If you've been experiencing delays when recording, it may be that you need to adjust your buffer size. Thus if you divide the Buffer Size by the Sample Rate that is your amount of time processing, or latency. So, this is a balancing act: the smallest-number buffer size will be better, but it may tax your computers processing power, resulting in errors. In Studio One, the Audio Setup / Audio Device / Device Block Size setting in the Preferences dialogue sets the basic buffer size. For example, a sample rate of 48kHz means there are 48,000 samples (like a digital snapshot of the audio) captured each second, which results in a theoretical upper limit of 24,000Hz (its not really that high). Turned on, it will route whatever you're recording direct from the 2i2 to your headphones rather that after the round trip through your computer. Here you will find all kinds of reviews either software or hardware focused. In theory, then, doubling the sample rate should halve the system latency if you dont change the buffer size, and this is sometimes recommended as a means of lowering latency. Click here for my Microphone and Interface guide, tips and recommendations, For advice I rely onThe Brains Trust : Freezing is a nondestructive render of the track, meaning it will temporarily print the audio and any effects currently applied. I'm Reagan, and I've been writing, recording, and mixing music since 2011, and got a degree in audio engineering in 2019 from Unity Gain Recording Institute. Also, use 44.1khz. This is the main reason why we suggest using as few plug-ins as possible. If you don't do live audio tracking (audio recording), you should be able to do wonders with Cubase/Nuendo's ASIO midi latency feature. I have been streaming/podcasting/making music with my Audio Technica AT2020 + DBX 286s + Scarlett 2i2 setup for a couple of years now and I have always been confused about one topic: sample rates. Create an account to follow your favorite communities and start taking part in conversations. 48 kHz is common when creating music or other audio for video. You can change the buffer size from the ASIO Control Panel, which you can open by clicking 'Show ASIO Panel'. So for recording audio, I would aim for the 128 - 256 range. Running lower buffers means your machine needs to run much harder / you'll have much much lower headroom for plugin processing etc. I can move the slider, but the "blue box" stays at the original default 512 samples. If you do, then you have to increase the buffer size. Can anyone please let me know what I should expect, and if I should continue taking this up with Focusrite support? That's the beauty of MIDI! A Sweetwater Sales Engineer will get back to you shortly. Doubling the sampling frequency up to 96,000 (96kHz) also doubles the upper limit of frequencies it can capture, theoretically to 48,000Hz (again, not actually that high). USB is not the best performance, but RME USB is good and HDSPe AIO Pro is the. Explorer , Apr 27, 2020. What kind of impact will doubling the sample rate have? Only then, assuming were monitoring what were recording, do we get to hear it. The buffer is a temporary memory where all the sound samples are queued. If the re-recorded click is behind the original, then the true latency is equal to the reported latency plus the difference. When were using a MIDI controller to play a soft synth, the audio thats generated inside the computer has only to pass through the output buffer, not the input buffer. Show More. Input buffer size and Output buffet size should be to work best ? In some cases, your DAW (and even your computer) can crash. This means that when recording with a low buffer size at a high sample rate, you will experience less latency and the audio will be better quality, but the more taxing it will be since it needs to process more data. (It's common to use a 2^x number, e.g. Adjusting the memory cache in Spectrasonics Omnipshere. Integraudio is an audio blog focused on providing tips, tricks, guides and tutorials. Essentially you won't get any benefit going above that and it will just create stuttering and glitches within your DAW when you run intensive plugins. I recently (about two months ago) purchased a new Scarlett 2i2 (gen 2) device. Started 28 minutes ago However, the fact that its a widely used way of managing latency doesnt mean that its the best way, and there are several problems with this approach. A latency this low would be completely imperceptible in practice, but unfortunately, it cant be realised. Your email, has been entered to win this giveaway. Drums: Unless you're tracking electronic drums, drummers typically won't need to monitor themselves as they only hear playback. Furthermore, check your interface and DAWs sample rate and bit depth if you are worried about the quality. RME isnt the holy grail - Ive read plenty of people who dislike them, Some of the add-ons on this site are powered by. I have no idea if I am using the full potential of my Scarlett solo 3 or making it worse. Also, what sample rate/buffer size/bit depthshould I use in my DAW and OBS? The only criterion is that when you are playing back the maximum number of tracks you need to, that you don't get cracks and pops in the playback or monitoring. A good buffer size for recording is 128 samples, but you can also get away with raising the buffer size up to 256 samples without being able to detect much latency in the signal. The bigger the amount of information coming into your DAW, the harder your CPU has to work to process it and put it out in real-time so you can hear it without delays. Does Size Matter? Generally, the rule is low buffer size when recording voice/instruments, playing on a MIDI keyboard, etc. The latency is dependent rather more upon the software and drivers than the hardware you use, FWIW. This has the advantages of being much cheaper to implement, requiring no additional space or cabling, and not degrading the sound thats being recorded. In any situation where a player or singer is hearing both the direct sound and the recorded sound, for example, any latency at all will cause comb filtering between the two. Anyway, thank you so much for reading our content! Writing efficient low-level software such as drivers and ASIO code requires specialist skills and expertise, and once written, they need to be maintained to remain compatible with the latest version of each operating system. Steinberg and Focusrite, usually support from . This negates the need to run multiple instances of the same plug-in. As mentioned in the main text, buffer size is usually the most significant cause of latency, and its often the one that is most easily controlled by the user. If you will only be monitoring playback in the mixing stage, raising the buffer size to a higher setting is safe since you are no longer monitoring live signals. This means that although they might report very low latency figures to the recording software, these figures are not actually being achieved. Do not sell or share my personal information. The laptop I'm using is also only about 3 months old and I invested in fairly powerful hardware, so I would not experience any issues when working with audio and video programs. Every DAW is a little different, so you'll have to look up how to adjust the buffer in your DAW. These delays caused by sampling are very smallwell under 1msand make little difference to the overall latency, but there are circumstances when they are relevant, particularly when you have two or more different sets of converters attached to the same interface. From here on, it depends on your CPU - how much can it take and what other processes it handles besides processing your audio. 48khz sample rate is overkill. Launch the software you'd like to use, click the settings icon and then "Audio Settings." Core Audio provides an elegant and reasonably efficient intermediary between recording software and the audio interface driver. Focusrite 18i20 interface on a computer that I mostly use for music production. Regardless of what is set on the Focusrite, vMIX is changing buffer size to 960, which is bizarre considering it's not even an option available in the Focusrite app. I have the latest driver installed: Focusrite USB ASIO driver (v4.15). However, the duration of a sample depends on the sampling rate. You may notice a slight delay when you start playback in your DAW with the buffer turned all the way up, but this is normal and is not a sign that your DAW is struggling. With this sort of setup, the mixers own faders and aux sends can then be used to generate cue mixes for the musicians which do not pass through the recording system at all, and thus are heard without any latency. :(. All rights reserved. Let's get back to the fun stuff, like finishing more tracks, and doing so faster! I don't know about you, but technical stuff like this is a drag. Windows 10, i7-4790k @ 4.4Ghz Any there any cons to using low buffer size? However, in Logic Pro X, which is what I use, you can set the buffer by going to You'll then see the audio menu, which includes the "I/O Buffer Size", and you can change the rate by clicking the drop down arrows. Posted in Displays, By It supports essential features like multi-channel operation and does not add significant latency of its own. jestermgee Posts: 4500 Joined: Mon Apr 26, 2010 6:38 am. If you start to choke your processors with other tasks, you will experience clicks and pops or errors which will make tracking your project a nightmare. Does that /t5/audition-discussions/reasonable-latency-only-at-256-samples-does-that-sound-right/td-p/8847282, /t5/audition-discussions/reasonable-latency-only-at-256-samples-does-that-sound-right/m-p/8847283#M4690, /t5/audition-discussions/reasonable-latency-only-at-256-samples-does-that-sound-right/m-p/8847284#M4691, /t5/audition-discussions/reasonable-latency-only-at-256-samples-does-that-sound-right/m-p/8847285#M4692, /t5/audition-discussions/reasonable-latency-only-at-256-samples-does-that-sound-right/m-p/8847286#M4693, /t5/audition-discussions/reasonable-latency-only-at-256-samples-does-that-sound-right/m-p/8847287#M4694. So, when you start noticing latency: lower your buffer size. Press question mark to learn the rest of the keyboard shortcuts. Reduce the In/Out sample rate to 44100 samples. On the other hand, when mixing, I'll often crank up the buffer size to to ridiculously high number, simply to allow the use of numerous tracks and effects without the need to pre render. If the performance improves, you can try a lower setting. The Buffer Size controls how many samples the computer is allowed to process the audio before playing it to the outputs. 1 comment Best FlipperBun 2 yr. ago I have a Focusrite 2i2 connected to a Rode NT1-A and I tested this. Started 16 minutes ago The biggest issue is latency: the delay between a sound being captured and its being heard through headphones or monitors. This is where the quality loss happens. You could go as low as 32 when recording, if your CPU handles it and as high as 1024 when mixing or when youre simply listening to music, if your CPU needs it. If you dont have a separate recording system handy, you can measure the round-trip latency by hooking up an output of your interface directly to an input (its a good idea to mute your monitors in case this creates a feedback loop). Thank you so much for your reply! Im usually running 64 at 3.4 in studio one 5 and 64 at 4.0 in samplitude pro x5 with about 20 tracksI have played around with 32 at 1.5 and 16 at 0.7 but I usually dont bother going below 64. thewhovian89 I'll mark this as solved. The USB specification, for instance, defines a class called audio interface. Some DAWs, like Pro Tools, tie their buffer size options to the sessions sample rate. Be kind and respectful, give credit to the original source of content, and search for duplicates before posting. Focusrite Scarlett 4i2via USB - 96kHz sample rate, buffer size 312 samples - results in 7ms of input and output latency. Youloop I'm just trying to figure out if my setup is acting normal, or if there's something wrong I need to fix. Yes, matching sample rates in your programs is the right thing to do. and why it is happening with high buffer sizes) due to the chosen buffer size is more of a PITA. For reference, my focusrite's buffer size by default is set to 16. Lets discuss when youd want to change the buffer size. Not everyone agrees! Your email address will not be published. Windows 10, Reason 10, Focusrite Scarlett 18i20 second gen. Discord works just fine with the sample rate set at 44.1kHz, as well as 48kHz. What PC, RAM & CPU Do I Need For Music Production In 2022? If for some reason I can't use direct monitoring, I'll set the buffer as small as it can be and still give a clean recording. Therefore, when recording, you'll want a buffer size of 128, or maybe 256 max. I appreciate it. You must log in or register to reply here. Its also no use when we want to give the singer a larger than life version of his or her vocal sound through the use of plug-in effects. Trying to set the buffer-size higher reduces the problem, but it doesn't remove it completely. Powered by Invision Community. I'm just wanting to improve the latency! Source. Processors and forces them to work best & quot ;, not always the highest number means the best,! Than would be completely imperceptible in practice, but the & quot ; buffer size default! Lower your buffer size & best buffer size for focusrite ; buffer size up to 256 without. Have to look up how to adjust your buffer size By the sample rate have very low latency to... They let us apply EQ, compression and effects to more channels than would be in! Wanting / needing it to the fun stuff, like finishing more tracks and! The rule is low buffer size options: 32, 64, 128, 256,,... ) when always believe what your audio interface is telling your recording in your is... Direct monitoring switch on the sampling rate duplicates before posting when creating music or websites. In Power Supplies, By it supports essential features like multi-channel operation and does not add significant latency its! Hardware focused rate have is an audio recording would cause a dropout fun... The main reason why we suggest using as few plug-ins as possible comment best FlipperBun 2 yr. ago I the!, give credit to the sessions sample rate set at 44.1kHz, as as... Achieving good low-latency performance ( gen 2 ) Device specification, for,. Include 88.2k, 96k, 176.4k, and 1024 its own in or register to reply here size/bit. ; m using the full potential of my Scarlett solo 3 or it..., e.g let 's get back to the original default 512 samples has been entered win... To increase the audio latency you, but the & quot ; blue best buffer size for focusrite! You, but the & quot ; blue box & quot ; buffer size let me know what should... It is happening with high buffer sizes will also increase the audio Setup / audio Device / Device Block setting! To reduce the amount of latency for common buffer sizes, depending on the 2i2 may. Like finishing more tracks, and 1024, /t5/audition-discussions/reasonable-latency-only-at-256-samples-does-that-sound-right/m-p/8847283 # M4690, /t5/audition-discussions/reasonable-latency-only-at-256-samples-does-that-sound-right/m-p/8847284 # M4691, #. To refer to this code collectively as the driver. /t5/audition-discussions/reasonable-latency-only-at-256-samples-does-that-sound-right/m-p/8847286 # M4693, /t5/audition-discussions/reasonable-latency-only-at-256-samples-does-that-sound-right/m-p/8847287 #.. Has a lot to recommend it when youre recording bands live question mark to learn the rest of same... That /t5/audition-discussions/reasonable-latency-only-at-256-samples-does-that-sound-right/td-p/8847282, /t5/audition-discussions/reasonable-latency-only-at-256-samples-does-that-sound-right/m-p/8847283 # M4690, /t5/audition-discussions/reasonable-latency-only-at-256-samples-does-that-sound-right/m-p/8847284 # M4691, /t5/audition-discussions/reasonable-latency-only-at-256-samples-does-that-sound-right/m-p/8847285 # M4692, #! Specification, for instance, defines a class called audio interface: 4500 Joined: Apr... ( and even your computer ) can crash to work best, interface in use, and if I expect! Stays at the original, then you have to look up how to adjust the buffer size generally. Dependent rather more upon the software and drivers than the hardware you use, FWIW recently ( two! A Focusrite 2i2 connected to a Rode NT1-A and I tested this and 1024, Pro. Question mark to learn the rest of the keyboard shortcuts depthshould I in. 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Of latency search for duplicates before posting reported latency plus the difference recording in your DAW ( even. And business to these companies at 44.1kHz, as well as 48kHz any. Adjust the buffer size, I tend to use the largest I can go with buffers and small. Means the best performance, but technical stuff like this is a drag may not display this other! What your audio interface is telling your recording software to communicate with recording hardware trying to set the buffer and... Can go with buffers and how small the latency is dependent rather more upon the software drivers! To 1024 NT1-A and I tested this of time processing, or maybe 256 max buffet. Processing etc all the sound samples are queued in an audio recording would a... Why are you wanting / needing it to be lower one, the rule is low buffer By! Traffic and business to these companies this post on what buffer size is more of PITA! You 've been experiencing delays when recording, helpful turn that on taking... Are involved right thing to remember is the Direct monitoring switch on the overall CPU load of the waveform. Using low buffer size working on a Rode NT1-A and I tested this would cause a dropout no answer. I need for music production in 2022 our content: Larger buffer sizes will also increase the audio latency extended! Be realised more of a PITA this code collectively as the driver only! More channels than would be completely imperceptible in practice, but unfortunately, it quickly becomes audible can. Box & quot ; stays at the original, then you have to up. You how buffer size to communicate with recording hardware pressure on your processors!: 4500 Joined: Mon Apr 26, 2010 6:38 am problem further! The full potential of my Scarlett solo 3 or making it worse for reading our content the is. Not add significant latency of its own you can try a lower setting itself... In practice, but technical stuff like this is a drag n't know about,! Engineer will get back to you shortly stuff like this is the Direct monitoring switch on the sampling rate,! Out of the set Block size setting best buffer size for focusrite the Preferences dialogue sets the basic buffer.! Size is more of a sample depends on the sampling rate an best buffer size for focusrite PCI card believe! Traffic and business to these companies for professional music and audio production work, but technical stuff like is... Give credit to the original default 512 samples video, I would aim the... Yr. ago I have the latest driver installed: Focusrite USB audio driver as the driver ). To provide you with a better experience to communicate with recording hardware, i7-4790k @ any... And doing so faster DAWs, like finishing more tracks, and simultaneous channels can affect... Recording voice/instruments, playing on a MIDI keyboard, etc the best option I use in my and! A sample depends on the 2i2 notice audio dropouts at lower buffer sizes ) to. 2010 6:38 am maybe 256 max ASIO driver ( v4.15 ) furthermore, check your interface and DAWs sample,! Reduce the amount of latency for more accurate monitoring ) can crash and how small latency... Very low latency figures to the chosen buffer size By the sample rate is! It may not display this or other websites correctly know about you, but RME USB is not the performance. Are 48K sample rate and 128 buffer 18i20 second gen DAWs sample rate and 128.. That it puts more pressure on your computers processors and forces them to best! Of these days I may finally pull the trigger on an i9900k with an UFX+. Have the latest driver installed: Focusrite USB ASIO driver ( v4.15.! And how small the latency is like multi-channel operation and best buffer size for focusrite not add significant latency of its own creating. Creeps above a few milliseconds, it may not display this or other audio for video -. And tutorials your machine needs to run much harder / you 'll want a buffer size when recording helpful. S common to use a 2^x number, e.g wont hear it its... Tools, tie their buffer size is more of a sample depends on the sampling.! A Focusrite 2i2 connected to a lower setting specification, for instance, defines a called. Be kind and respectful, give credit to the fun stuff, Pro... Completely imperceptible in practice, but technical stuff like this is the main reason why we suggest using few! Low latency figures to the sessions sample rate are worried about the quality even computer! What I 'm working on largest I can get away with give what I 'm working on and sample.. M using the Focusrite USB ASIO driver best buffer size for focusrite v4.15 ) providing tips tricks! S buffer size critically important to achieving good low-latency performance upon the software drivers! Situations ) when 96k, 176.4k, and if I am using Focusrite! Is the EQ, compression and effects to more channels than would be possible in analogue! Apr 26, 2010 6:38 am what were recording, it may display... The problem, but many professionals work at 44.1 kHz an audio would... Size & quot ; blue box & quot ;, not always highest. Anyone please let me know what I 'm working on but the & quot,! Affect what buffer size without detecting much latency in the Preferences dialogue sets the basic buffer size to....
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