WHAT is the BEST SAMPLE RATE?
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- Опубликовано: 26 янв 2021
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i find 0 Hz produces a really warm tone that you cant replicate with higher sample rates
NICE ANALOG WARMTH
432 is much better. It can potentially heal cancer.
@@owenathanael The sad thing is some people believe that ;)
@@iliatilev true.
#noTroughsHereGang
You are a person in the audio world I really respect and also the last one to say that recording at 48 kHz makes sense, I remember your old video where you said that 192 kHz was the best. To be able to rectify your ideas is praiseworthy. Cheers!
high sample rate is really useful for resampling in sound design tbh, its nice having that high head room when you pitch stuff down many octaves down or really on weird distortion and other audio mangling
This is true, it's very noticeable.
Ahh, I see this is makes sense now
I've worked over the years at Universal, Enterprise, many big studios and the most common preferred bit and sample rate is 48K 24Bit, so i've always worked at these rates plus the plugins function much better unless you have an HD system
Video editing programs call for 48k/24bit and so I have always used that as well for the compatibility. Set it and forget it.
Though I’m not a mix engineer. I’m a film composer and songwriter that does his own tracking, and I am working with a lot of samples that are already 48k/24bit as well. It’s pretty much the standard in my world.
No this is just solely music production
@Lostie DeMonde Absolutely, bigger numbers are always a great selling point.
By far the most insightfull video I have ever watched upon the topic is the one on the FabFilter RUclips channel "Samplerates: the higher the better, right?". Very interesting video, very much een aanrader!
Nice to see you talk about this too! I made a video about this way back, and made sound examples with some cheaper converters. Some people got pretty upset and claimed that I did something wrong haha.
Would you mind directing me to that vid?
I have always worked at 48khz for many years, pretty much for the same reasons you explain. Well explained as well!
Absolutely love your videos!!! Great advice. Hats off!
I went to 48k from 96k about a year ago for your reasons and also to save CPU overhead and drive space. It's been great!!!
192k yields some unruly file sizes to catalog, really only noticed a difference with a client playing classical music on a 9’ Grand piano with tremendous dynamic range in the music. Also it greatly reduces your access to how many plug ins can be instantiated. I agree-24bit/48k is the ideal tracking paradigm.
Stock plugins are good enough, also manufacturers can always upsample their plugins
How can the dynamic range be affected by the sampling rate? The bit depth controls the dynamic range, by lowering the noise floor. And the hardware, no matter if it’s audiophile consumer hardware, a prosumer audio interface or professional gear, has yet to surpass the limits of a bit depth of 24 bits. And nobody would ever hear a noise floor of even 16 bit recording, unless the volume is at a level that will hurt and degrade your hearing capabilities quickly. So the advantages of 24 bits really only are of any interest in the field of recording and mixing. Or do I forget something here?
192k has nothing to do with the dynamic range, it’s bit depth that is responsible for dynamic range.
>really only noticed a difference
stop lying mxr
Good video! Totally agree, samplerate above 48 kHz is redundant unless you gonna do some pretty gnarly sound design, purely to reduce aliasing across the board. Really easy to resample to 48 too in a good DAW that has a focus on hectic processing. Something like a Reese bass with LOADS of saturation or a very smooth sounding FM bass that has multiple stages of frequency shifting (basically the situation described at the end of the video)
Very very cool. I was just thinking and messing around with this last night.
My Tascam maxes out at 48/24, so I'll vote for that... Second place would be a Maxell xll-ii running at double speed 🤷♂️ ;)
A very clever idea is that you let the one file run in 192 khz and record this file with 48 khz so there will be no processing of the rate, its more like a recording of the configuration, without changing it on sonical levels. This is one of the best solutions.
Always been a 48 kHz guy myself
6:08 it is worth putting reaper on the extreme setting for sample rate conversion. It has the least amount of aliasing according to that sample rate converter comparison website which I don't think I can link to without RUclips dumping this comment in the spam folder.
Good video, Wytse!
finaly a Good explanation why you should(Could) cHOOSE for 48khz over 44khz . lots of people in youtube videos say:just work in samplerate whatever the final file sample rate will be. because of the fact that the conversion from one samplerate to the other could possibly create new problems. but very clear why u choose for 48khz. thats why im subscriber i need all this nerdy stuff. Keep on going!
I have had the option of high sample rates for a long time, but the final format for the film scores I worked on was always 48k, 24 bit. My decision was to track at the sample rate of the final (dub stage), so I recorded and mixed everything at 48/24. My experience is that Sample Rate Conversion does much more damage to the sound than any benefit that may be gained using high sample rates. The 3 worst words in digital audio: Sample Rate Conversion.
Your Mileage May Differ :)
I run Studio One at 48kHz. The problem is that the rest of the world is typically 44.1kHz. When I leave Studio One and go to say, Soundcloud, I have to remember to change the Apollo to 44.1kHz. Maybe I'm just doing something stupid....it wouldn't be the first time.
Thanks for the great video! When you said: Some plugins have it and some doesn't because they don't need it - can you explain which type of processing needs it and which type doesn't? Like distortion, eq, ... ...
At the end of the day, its somewhat system dependent - recording 96k sounds a bit better than 48k on my system, but I do not know if that is plug in related and I've not been able to tell the difference on other systems.
I used to run at 48KHz, but recently moved to 96KHz since I now have a much more powerful computer.
Pretty much all plugins support it now and any that do not internally oversample will benefit from 96KHz natively due to the reduced aliasing. Wont make any difference to those that do already oversample though.
What format do you convert to for 96
@@WaterRiver777 WAV
@@insanebiscuit1 where do you upload too
@@WaterRiver777 i dont upload, i mix for others
The Resampling feature at 6:00 is basically how accurate you want sample stretching to be. When you stretch a sample to be a different length than the original, you might run into these really digitized, computery artifacts (especially if you make the sample slower). The higher the resampling the better the stretching will be, and the less audible the artifacts.
That computery artefact is so nice of crunch to samples. Thats the whole reason i now use arturias emu sampler. 20 years back used the emu 1212 chipset for that.
Stretching is a concept not quite applicable to samples. They are defined as scaled dirac impulses which have a infinitesimally small width and cover a area of exactly 1. The option in reaper fives away all the information: 512pt sinc interpolation. You use a sinc function sin(x)/x where x is chosen according to the desired low pass frequency. A perfect sinc goes from -infinity to infinity though. You approximate it with 512 points here and call it good enough. The 512 points (if one side is meant) lead to 20*log10(1/512) = -96 dB which is the theoretical limit of 16 bit PCM. For each resulting sample you need to do 512*2+1 evaluations of the sine and one division though, both quite expensive operations on CPUs. The sum over all is cheap at the end. The amazing part about this technique is that you can choose any resulting samplerate you want. Fancy some 37kHz ? No problem.
TLDR; samples can't really be stretched, sinc interpolation is great but computationally expensive.
@@Mefistofy I meant an audio file, not individual samples, sorry if it wasn't clear. Stretching something like a guitar to last longer, or be shorter. They are indeed pretty expensive on CPU, and that's why one should save higher point counts for offline rendering.
@@iammodus Did not think of the sample as sound recordings. I might bee a little to much into DSP sometimes. Everything you mentioned is exactly on point.
An additional consideration when choosing converter sample rate is latency. If you're doing "offline" work (record first, then work with the result) you may not care, but it can be quite impactful for "online" work where you want to minimize the delay between input at the microphone or instrument and sound out of the speaker (of course, converter rate is not the only reason for latency, just one of the multiple factors in the equation)
Do you remember I was talking about this, that some people were hearing the intermodulation distortion from the convertors, as an improvement in the treble response (One day you might get a plugin to give it that vintage I.D. sound, though I hope not!)
I have been running projects in 88.2 with my 80s analog console...
I was told that if you plan to sum or do external processing that it’s better to leave digital domain at highest rate possible.
Any truth to that??
I track at 48/24, master is tape and then 96/24. The quality of the converters is more important that the sample rate. Depending on their clock and design, converters can sound better at specific sample rates. Find your sweet spot.
do you ever have issues converting a client's 44.1 khz project to 48 khz?
@@johnthecreative I would keep it at 44.1 while mixing
@@Synth2000 tell that to many pros that do 48.
@@johnthecreative My 44.1 comment was a response to the former poster. I currently work using 88.2 and DSD, and for many years I was on 48/24.
I'm trying to think of if there are any hangups with this workflow when working with soft instruments. Do they have to internally upsample by default like u-he?
This is really informative! Of course Reaper's way of handling "internal oversampling" is quite unique I think! Is there a way to do this in Cubase??? You can allow audio import with multiple sample rates in a project. But I am not sure how Cubase then handles the whole oversampling, aliasing etc.
By just working in the box I like the precision and the less harsh highs (treble) in the rendered master.
yeeeeees thx white sea studio i also work at 96khz and export at 192khz for aliasing and to apply oversampling built-in or use metaplugin for aliasing and convert it back to 48khz using dbpoweramp which is the most clean pristine sample rate converter available on the market.
Higher sample rates on conversion are useful for creative sampling - if you plan to radically slow down or pitch down your samples, its a good idea to convert at higher rates, so you do not lose all the high frequencies.
Actually it does not work like that. If you tune down your sample, your low frequencies get pushed to high frequencies area, so your 100 hz becomes 200 hz, etc. So having frequencies above 20000 hz won't help - they will be translated to the inaudible frequency range anyway.
@@kadavr0s I think you got it backwards. Maybe I've explained it poorly.
@@kadavr0sNah fam, there's definatley a huge difference in resolution if you're slowing down or speeding up high sample rate files. It's almost like recording at a high frame rate with a camera.
OK, holy crap.
I just took an old mixing session I did at 44.1kHz (because all those audio files are 44.1kHz) and just increased the project sample rate (in Reaper) to 192kHz, and holy crap, the dynamics improved like crazy...!
I did nothing else, just increased the project sample rate to max.
So, I did some quick testing.
1. I rendered a fresh 44.1kHz version, with the project at 44.1kHz. (Just to make sure it's this exact mix I'm listening to.)
2. Then I rendered a 192kHz version, simply setting the project to 192kHz.
3. Then I loaded this 192kHz render and put a brick-wall lowpass at around 20kHz (analyzer showed some action above that, so I wanted to manually remove it), and rendered it back to 44.1kHz.
Versions 2 and 3 sound more or less the same, with very much improved dynamics, especially the snare. Simply allowing it to process in 192kHz, even after later downsampling back to 44.1kHz (native resolution of the audio files), made a huge, huge difference.
Obviously, the true 192kHz version has a tiny bit better quality in reverbs and such, but that's not the point, here. The point is that the dynamics improved so much that the mix would need some adjustments to adapt to this change -- it's not a subtle difference at all!
The snare now gives me a headache, as it seems to shoot out from the mix and punch my forehead...
The upper range of the distorted guitars is also more defined in versions 2 and 3, though this is more subtle.
(All renders were still native 24bit, and I used lossless FLAC format. Only the sample rates were changed as stated above.)
Hi, thanks for the video. I have 2 MOTU 16A units which I intend to run at 192kHz on a mac. I noticed that you are using a mac mini at 192kHz and wondered how many audio tracks would run on the mac mini. I want to use it like a 32 track multitrack tape recorder with some basic plugins like compressor and eq with an occasional effect on each channel. I could also do with very low latency as I am very sensitive to timing issues when overdubbing. I am not sure if an M1 mac mini with 16G of ram would do the job or if I should get a mac studio max with 32G of ram. You are one of very few people on youtube who are running at 192kHz and I wondered if you could help me please? Thanks, Kev
The one big reason to use 192khz is if you want to time stretch audio. More sample points, better result.
Yes there are more sample points in general but the region,say,between 20hz-20 khz is represented with the same number of sample points in any scenario.
The link of the "copy" of the article direct me to a Chinese website? is it the right link?
i got a question, so if im using waves plugins, can i just record in 192 and then process the waves plugins without that nasty noise distortion it creates and then finalize it and then back to 44.1 will it sound way better? since waves has no oversampling
It would be nice to know what plugins and soft synths etc oversample/upsample. I know UAD does, but Acustica Audio requires that the audio project is in the higher sample rate to use that version of the plugin. What about others? A list would be good.
The link is invaluable. Many thanks.
2:00 In fact, the reverse claim is also true. Some more or less ancient low- to mid-end interfaces only deliver good sounding monitoring when forced to operate at high SR. Everything now relies on delta-sigma and a proper digital filter algorithm should do the job, but some devices have a nasty roll-off in the audible part of the higher frequency range.
Is there a way to process a Pro Tools sesion at a higher sample rate like shown on reaper in this video?
Your favorite plugin manufacturer Acustica Audio recommends running their plugins @96k :)
what about sound design? aren´t higher sample rates more interesting there? i mean, in the terms of audio-stretching, slowing things down and stuff like that. (i am talking digital games and scifi movies here). in theory, you should be able to slow audio down without hearing too much artifacts when recorded in higher sample rates, right? (and you are golden when you have enough money to buy those sanken mics that go up to 100kHz)
Yes and no. No to the fact that a band-limited nature of sampled audio it will not help you extrapolate audio because, the signal is fully defined by the samples. However, higher sample for reconstruction and noise reduction higher sample rates help because, clips and pop can easily extend to 50kHz and do build to filters to remove them you need that content to exist.
Been working at 24bit 96k since 2004. I hear a significant difference switching down to 48k, and only a slight difference moving from 96k>192k. 192k may very well kill my ancient rig.
@Joe Smith In theory, you may very well be correct. If you get a chance...try it on the old RME FF800. We hear the difference over here in the studio.
@Joe Smith most times yes as we track at 24/96. So provided the source files are recorded in high quality then yes. I'm not here to debate as I'm snowed under with radio work and need to prep for tomorrows show. Each to their own
Its a simple enough test..play a high quality Ultra HD song and from RME's Fireface settings switch between the sample rates as the song is playing. You should hear it for yourself (unless I have a defective unit of course)
The vids are getting better! 96/48 here. 48 is "common ground" and "plays nicely with others".
I’m confused. I work in logic, and if I record at 48 and then change my project settings to96 or 192 after the fact it changes the speed of the playback.
Plus, I’m not really understanding how you can process at 192 when you recorded at 48. Where is the computer getting the extra data from if it wasn’t recorded in?
Only time I find use for higher samplerates (96kHz), is when I do field recording-I do walk around with mic and headphones and capture interesting sounds, I can later transform in studio. Samples recorded at higer samplerates respond, sound better, when time stretched (usualy slowed down). And later do bounce them to my working asmplerate-48kHz, that I also use for recording bands.
Would be interesting to record some music at 192Khz and then spectral shit the top part into the audible range (without folding like in aliasing), would be the equivalent of a gamma ray picture.
This guy is so cool and he has such great knowledge about music production, hardware and sh1t that I would be happy to have such a friend. it is like adding saturation into my life.
Heya! I wanted to ask what’s your approach with sample rates, as 44.1khz is standard for streaming services, but 48khz or more is usually for video/film. Do you downsample for streaming services when working with 48khz? Thanks!!
I dont know much about those numbers in the exporting box but On FL studio the 'highest' setting of it is the 512 one.
If anyone knows can pro tools do the rendering of files at 192 while the session runs at 48 or is that only reaper?
The link for the article copy is not working ! can you re up please.
I've been recording and mixing at 96k with an I9 20 core 48g Ram setup using reaper as my DAW. I'm currently using Presonus CLARETT for my interface.
I find that large track and plugin counts are giving me lagging issues.
I've streamlined the buffering settings in reaper for my rig but still having issues from time to time.... You mentioned a testing website to see if my converters could handle this... But the only links that work for articles on your page send me to a Japanese website...
Any suggestions?
Thx, Jeff
I use 3 audio interfaces so I don't have to switch between 44.1/48 or greater sample rates. I feel 24bit/48khz is perfect for videos and most applications. When I hire a client for professional vocal audio-work, I'm perfectly happy with receiving studio-grade 24bit/48khz audio files.
Would it make sense to use a higher sample rate when multiple roundtrips are required? Say if you decided to mix fully analog, but needed commit and re-record tracks along the way due to a lack of hardware? (So say I want to process a bass guitar with several compressors, but I only own one, so I would re-record the sound twice, before sending it out to the hardware again a third time.) Would there be less information lost if I say used 192 for the first two, and then recorded the final mix as 48, or 44.1K?
5:59 resample mode seems to be a way to select which interpolation method is used to convert from one to another sample rate. check out the wikipedia article of sinc filter for example to read about the best ones in this list. little spoiler: the more points the sinc filter has the steeper the anti-aliasing filter can be.
btw i think it was a good solution to just tell your client to send you everything at 192khz, because didn't have to explain anything that way and just had to wait for new files. but essentially the real problem is probably much more that the client doesn't know how to use oversampling features and it would be better for the client in the long run to just learn how to do that.
iLOVE Your videos a lot you are a source of knowledge Thank you
So, if I record my field recordings at a really high sample rate, can I also slow them down without getting quality loss, or gaps in the audio?
I just record podcast and apply basic fix like compression eq . I use Zoom H4n pro . Will there be significant difference between 48 khz 16 bit vs 96khz 24bit?
Can you make a video about audio 32-bit float depth vs 24 bit?
Good info. Thank you very much!
i work at 44.1 whenever possible and disable oversampling because I enjoy the sound of aliasing. I often also add bitcrushers on delays and reverbs to crunch them down to 22kHz for extra grit.
@espoir inconscient happy hardcore, jungle, nu-metal, and generally anything that takes my fancy. i tend to blend together elements that aren't usually seen together like mixing new-age pads with bluegrass banjo or such.
Over the years I have found that upping the bit depth from 16 to 24, sounds better than doubling the sample rate. Also, converting from 48k to 44.1k sounds better than going the other way. I can also see the advantage of plugins, even mixing itself at higher sample rates. So, I record at 2448 and let my tools up sample if they sound better and live with it. But that's only my opinion!
The measurable improvement from going 16 to 24 is VERY noticeable. I always tell my clients that if they can improve one thing with their recording chain, it's that.
I preffer to use plugins with oversampling on 44.1/48 not only because my CPU doesn't need to struggle as much, but also because of compatibility.
hello.please tell me what kind of CPU is needed approximately to work in 192? please advise.
Hey Wytse. thanks for some always great content :) The alternative link you provided for the article seems to lead to a chinese unsecure website. Just so you know...
What I noticed is that is seems to me that the Better mode when rendering is better for uploads on the web. I at least had issues when putting my projects on soundcoud when using the extreme mode
allow me one question: what format would you expect from your customer to work with, and what format would you give back to the customer? Is it enough to get 48khz/24bit and give back the same format?
96k/24bit is what I use and I like it. I record into my desktop because my pos laptop cant keep up like you said.
The link is not working (japanese or chinese web site)
link does not work. anyway i'm trying to run my system at 96khz and it runs ok.
Just curious... Is your new M1 Mac Mini handling the 192 sample rates well?
I run on 88.2k, as it's the easiest for the converters to handle (to 44.1k) , unless there are 2 dedicated circuits for (doubling) 44.1 and 48k. 192k should be converted back to 48k if you want to be sure of matching bits and their calculations. Floating integer calculation at 64-bit will handle the conversion better, but I would stick to doubling/ halving myself.
It seems like it should be easier, but mathematically the conversion from 88.2 to 44.1 is just as complicated as 96. It's not a case of removing bits or samples or anything, it's still fourier transforms and curve fitting.
@@EricOehler01 as far as I have heard dedicated to multiples was the way to go. It seems logical too when dividing 2 by 3 in integers. that's where the 'curve' fitting gets slightly affected, The bits are changed by the tiniest float points or something like that. I saw it in a presentation from Marantz in their high end stuff. It def changes the sound, butm at the moment processing is way improved, so I wouldn't worry that about it that much personally. That said, I record and mix in the same samplerate. Always. If I change the settings on both my interface or DAC there will be differences in the sound. And not always for the better. Steady is the way to go imho.
@@EricOehler01 I thought the calculations are easier with 88.2 to 44.1 and vice versa because no interpolation needs to occur between samples. That vs something like 96KHz to 44.1 which would require some interpolation because of the non-integer division of the sample rate no?
Eric is right here. The “easier calculations” thing is really a myth. I always encourage people to do their own tests to see what actually works and sounds better. So I encourage you to try 96 vs 88.2 and see what you like best. At those rates it’s a slight difference and probably doesn’t matter very much though.
@@Dthebeatsmith it always requires interpolation, because downsampling isn’t strictly a divisive operation. My DSP math is weak and I haven’t done any since college a loooong time ago, but YOURE not just picking the path between samples, you’re refitting a curve to a new periodic rate. Basically saying “this equation described this curve at x sample rate, what equation does it at y”
We’re used to the notion of halving and multiplying sample rates because it makes logical sense, and works for visual mediums and frame rates. But the math is complicated and fussy.
If it were as straightforward as halving things, there wouldn’t be any differences between SRCs and we’d never have to worry about whose DAW did the best downsampling. :)
in ableton my latency is lower with 48000 lower than 44. how is that? :)
I didn't understand a word you just said but it sounds very interesting lmao
Hahaha this was me two years ago! Keep with the hobby
It shounds very intreshting
Domagoj Vida changed profession, from football to producer 😎 P.S. Tnx for video 🙂
I tend export wav files from Korg gadget in 24 bit / 96 kHz because it's the highest it can export but I'm wondering if I should just do 48 kHz to reduce file sizes as those 96 Khz get pretty big. When I import them into Cubasis it always warns me that it takes more cpu load on the iPad. Seems like the audiochip in the iPad can't go any higher then 48 kHz for playback unless I connect an interface to it. On the PC side my interface (Steinberg UR-242) Can do 192 kHz but I almost never use it that high. I do the mixing with both Cubasis and Cubase. I do the initial rough mix on the iPad with Cubasis and import the project into Cubase on the PC to tweak / improve / export it. Those 96 kHz files take up a lot of disk space in regards to backups. Maybe 48 kHz would be a wise idea next time..... I do use 24 bits though, the projects involve only electronic sounds generated by soft synths and are instrumental with only some stock plugins.
I write electronic music at 96khz because I can get lower latency on soft synths, at the cost of more CPU. For mixing purposes 48khz with plug-in oversampling like you said it's the way to go.
Quality is often the decimal precision (how many digits behind the comma are used to describe the wave). Higher quality means that the algo needs to calculate much more to come to better precision. The opposite of it, is cutting away precision. This is also, what a bitcrusher algo does.
Do you convert your 48khz projects to 44.1khz after finishing?
I had designed a guitar tone I really liked using a couple of different plug-ins. Then I decided to go from 44.1 to 192 in Logic. Suddenly my guitars sounded horrible. I did tests at 44, 48, 96 and 192. The difference between 44 and 48 was subtle, but overall it seemed the higher the sample rate, the worse the tone. I decided rather than redesigning my guitar sound, I would just stick to 44.1 for now. But still not sure why this is the case.
Food for thought!!!........... Thanks
damn steven wilson thanks a lot! you got yourself a new sub :)
the links to the article are broken :-(
Pardon my ignorance, but how does one tune a guitar to 432Khz? I assume an electronic tuner of some sort is required. I'd like to try this.
At what sample rate should we record audio? Thanks
Good afternoon,Mr De Roote, I am waiting for your UVI Shade vs Fabfilter Pro Q and Saturn
i can mix longer when on 44.1.... sometimes i switch to 48 but i keep getting back to 44.
1:34 But I thought that the sigma delta converters that we use for audio didn't need a steep analog antialiasing filter, since it's the digital FIR filter that does the "real" antialiasing...
If you got oversampling on the plugins where aliasing is a problem like Saturation, Distortion etc, 48 kHz is enough. You don't need any more.
Spot on!
Very interesting. I would like to do this test, that you mentioned in the video, but the links in the description don't seem to work. Could you update them, please? Thank you very much!
@Simpler Sounds Ah, I almost forgot about this one. Read the article, that's linked in the description and play the mentioned files on your system. The files should be completely silent, but if you can hear some cracks or noise, your system isn't capable of handling high res audio.
@Simpler Sounds Yes, it must be maxed out. Otherwise the hardware doesn't play these ultrasound waves, that lie above the human hearing range.
I just tested this with my on board sound card. I get some slight cracking noises at 192kHz, but none at 48kHz, because the low pass filter cuts those frequencies out.
I record crazy modular synth patches at the highest possible sample rate as the modular synth makes noises well out of the range of human hearing and when I střech the audio and pitch it down I find sounds I wouldn't get at the normal sample rate
Also 24/48 in the mastering room and production room here.
What about the bit rate?
great vid ,very interesting
none of the article links works?
So for gaming,is the 48khz sample rate is the best choice?Just curious.
Thanks for explaining that feature in Reaper! Finally I know what this is all about.
So after having worked with Reaper a longer time are you happy with Reaper?
I went from logic to reaper for mixing and mastering. The ability to do processing in 96k but use 48k files seems to be unique. Works like a Charm and sounds amazing.
been running at 88 for a couple years now but likely going back to 48 for the next projects and onward.
do you ever have issues converting a client's 44.1 khz project to 48 khz?