There are plenty of bad sample rates converters that unbalance the sound when down sampling. So imagine the conversion algorithms of the streaming platforms, there must be some real bad ones out there too. It is better, to avoid all these problems, to always upload music using a sample rate of 44.1k to any streaming platform. So as explained in the video by Streaky, a dedicated specialized sample rate conversion tool is recommended, because most of the ones provided with DAWS are crap and unbalance your sound when performing this down sampling conversion. Also, think about this as an engineer when delivering music to an artist: the down sampling conversions that the artist may do with the music that you send to him (for example, when he makes a video clip for RUclips and inserts the sound track into it) may alter the good sound that you made. Make sure to recommend to all artists to only distribute and use the 44.1kHz version and make sure that YOU make it, not them. Otherwise you may see your name on a production that does not sound like the one you made. Best Regards from your friend François in Montreal, Canada! 🎼🃏🎶🎵🙏😁👋🌠
@@queenpurple8433 A vast majority of high-resolution music files have a sampling rate of 96kHz or 192kHz, both of which are appreciably higher than the 44.1kHz cap on other digital files. High-resolution streaming services allow you to stream music at 96kHz or 192kHz if you have the right gear.. so it may be a good option as well to proceed with your method if you want these platforms to play in your high-res sample rate. - Apple Music, Spotify, TIDAL and other online stores/streaming services - 16-bit/44.1k WAV files - Apple Digital Masters - 24-bit/96k, 88.2k, 48k, or 44.1k sample rate WAV files - Bandcamp and SoundCloud - 24-bit WAV files (sample rates above 44.1k) - Vinyl - 24-bit WAV Files (sample rates higher than 44.1k if available) - Cassette - 16-bit WAV Files (sample rates higher than 44.1k in some cases) - RUclips - 24-bit/48k WAV files or 24-bit/44.1k Be mindful of the delivery specs for RUclips (or whatever video platform you are using) when the video editor makes their final export. Avoid using mediocre sample rate conversion algorithms that are often found in video editing software, or basic DAWs like Pro Tools, Logic etc. iZotope RX has great sample rate conversion, and most mastering engineers have “mastering grade” sample rate conversion software which can produce better results. I personally use SOX to make the sample rates conversions and my sound keeps a 100% perceived fidelity when making these conversions and comparing both with high-definition headphones Because your audio for video is likely to encounter some sample rate conversions and data-compression, many recommend leaving at least a full dB of peak-headroom to avoid any clipping due to the conversion processes.
@@themagicianofsound so it's better to downsampling the file before sending to streaming platforms? I would like to know more about this, in my experience I've always upload at 48khz and I've never encountered problems btw
@@AftertuneMusic If you are happy with the results, I would say that you can keep going with the method that you are using. I am more concerned when I give music to artists (especially non-professionals) who then upload on various platforms and also make their own video clip or their own lyrics videos. They use various tools and various conversion methods without knowing what they are doing. It already happened a few times to me that I was hearing my master on their RUclips channel and that the EQ balance was not exactly the same as the master that I delivered (usually the most perceptible symptoms are lighter bass between 63 and 160Hz and higher mids between 2k and 3.5k). After making extensive research and testing, I was able to reproduce the problem myself and this EQ unbalance was happening when down sampling or up sample the sample rate with various DAWs including mine. I now use SOX to perform sample rates conversions and not my DAW. This allows me to keep a perfect sound when performing these conversions. One of the music producers I work with was having the same problem with another DAW when performing sample rate conversions and never noticed the problem until I tell them "Hey wait, your 44,100Hz and 48,000Hz versions have slight differences in the EQ balance that I can perceive". So I prefer now to give to artists a 44,100Hz 24 bits version that I create myself and I specify in the filename "for streaming platforms and RUclips" and another version in 48,000Hz 16 bits and I indicate in the file name "for SoundCloud". So I am sure that the artist or the upload process will not mess up the EQ balance because I am very sensitive to this. But if you are uploading your own music and you feel that your sound is 100% accurate when your music is uploaded, then there is no reason to change your process in my opinion.
@@themagicianofsound thanks a lot for the info, it's always nice to have more opinion in this tecnical issue. I personally now produce and mix at 48khz then i mastered in another session at 96khz and then uploading, but now I want to check more about balance shift.
I learned that it sounds ok if you divide or multiply by 2 like 88.2 to 44.1 because it's just half or double the samples, but you run into issues if you have to convert from 48 to 44 for instance. Why didn't you say anything about that? Isn't this a big issue anymore with a good converter? It's hard to imagine that if I do the math. I would like to know because this is the reason for music cd's and severses that I go for 88.2 or 44.1. Thanks
Because it's nonsense. It's not the division that's a challenge when converting sample rates. It's the filtering. If you'd just throw away every other sample when converting from 88.2kHz to 44.1kHz you'd be using the most naive and worst sounding conversion possible, with any signal above 22050Hz folding straight back into its aliased equivalent frequency.
@@michaelanderwald4179 That makes sense. Thank you for this explanation. An engineer that I respect for his excellent work explained the double or half rule. But of course, when you take every other sample you get aliasing.
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44.1 kHz and you are done. There is no reason to even record in anything higher kHz. Why? Several reasons. First off: Did you ever hear about the Nyquist-Shannon sampling theorem? Strongly simplified, it proved, that a sampling rate of twice of the highest frequency you want to record is enough to create a perfect digital audio sample. According to this theorem, you can create a perfect range of 1-22050 hz. With any sort of post processing possibly needed you will get this outcome out of a source material at 48kHz of sampling rate. Also, on most audio interfaces, the recording gets worse the higher of a sampling rate you’re trying to record. Even on most prosumer grade audio interfaces or recording devices, 48kHz will sound pristine, while at 96 kHz, recordings will greatly suffer, let alone 192 kHz.
Mate, the higher the sample rate the better recording audio. processing is more accurate. Level meters respond more accurate and thus anything that reads level ie. compressor is more accurate. Distortion and Saturation bounce back with much less Aliasing that effects the audio high and low ends. There's a reason best plugins have oversampling options; they run at a higher sample rate the convert back down to session at say 44.1 kHz. And Im not sure what interface you are using but my simple SSL 2 sounds much better at 96khz that 48khz. The reason I use 96khz is because I mostly use Waves and they don't have an oversampling option..
that's a theorem. in practice you need a perfect filter that will reject all aliasing above niquist, very hard to build. hence, increasing the sample rate will move errors/ aliasing into inaudible spectrum . apparently 48khz willdo the trick, yet 60khz is perfect yet nowhere to be found in common recorders. so, 96khz it is!
Please note that You can’t downsample from 96khz to 44.1khz. They are not compatible ! and will alter the sound… recording for 44.1khz must be made in 88.2khz and For video format in 48khz you should record at 96khz.
@@nicksterj i didn’t say it s not possible. It’s not compatible. I suggest you Do some research on 44.1khz and 48khz . You will altered your source sound going from one to the other. They are not compatible
I have a hard time hearing improvements over 44! maybe because of the kit I use or maybe because I have listened to so much at 44. so much music I loved was made at 44.
@@380stroker have you found a solution that’s better?I’ve used both RX and R8Brain with great success for batch processing: I’ve also used the old school analog way of playing back each tune in real-time and converting sample rate between two converters and separate computers … but that’s not efficient in time management…. Please let me know which other solution has been working for you for batch processing… I’d love to give it a go
@@ZiwaHD For the sake of the client, its always quality over quantity. I don't care how long it takes as long as the end product is optimal. R8brain is horrible. Seriously.
Distortion is the main thing that produces high frequencies. So while I'd have no qualms about doing classical music at 44kHz, I'd really prefer higher sample rates for distorted stuff, especially when I'm actually producing the distortion in the digital domain.
About 88,2 Khz? I use this sampling frequency because, like Bob Katz says, is perfect and gives less errors in 44,1 Khz downsampling .What do you think?
That's actually incorrect (sorry Bob). If you do some research, sine waves are drawn the same, as long as they are not above the nyquist frequency. I find 96k to be a good balance of I/O channel count, cpu power use, and headroom (still should use 4x oversampling for saturation and certain other plugins when possible).
Ah yes, Bob K and his K-system. Look, sample at whatever rate you want. Just make sure at the end you use hardware instead of software to do your last conversion.
96 Khz is way overkill for just producing songs in the box. The only difference you will notice is the more than double load on the CPU, meaning much less plug-ins. It's a total and unjustifiable waste of CPU-resources. It's gonna end up in 44.1KHz/16 bit anyway and, as said, when needed vst-i:s will upsample (higher samplereate) internally and the DAW already handles things with 32 or even 64 bits internally. So everything that is important gets the "higher treatment" already WITHOUT the extra CPU-load overall for every plug-in. Especially if only working with non-acoustic stuff. Mastering acoustic music may be a totally different thing that I don't know anything about though. But checking about A/B blindtests people can't hear a difference when you go above 44.1/16 bits. There's a reason why it was chosen in the first place for the CD-standard.
You mentioned one VERY important point - I mainly do live recording of bands and orchestras and ONLY record at 96k to capture a good hi resolution signal. If you are working in the box on EDM etc with samples that were created at 44.1 or have come from mp3 then sure - there is no point in working above 44.1k
@@hanslutz6420 My soundcard will get 2 ms latency at 48 samples @44.1 kHz(1 ms in and 1 ms out), which is more than enough for anything. I have it set at 256 samples buffer right now...
Hi, one slightly basic home recordist question. Would you advise I get a multi effects guitar pedal that outputs at 44.1 kHz or one that outputs at 48 kHz. Like you mentioned, audio typically requires 44.1 khz and video requires 48 khz. Since I intend to use the unit on both types of projects, would upsampling the guitar parts be better, or downsampling? As work flow goes I’d need to record it in the device’s native sampling rate, then convert it into the project’s sample rate, and then import it into the project - would there be any better way to do this. Of course, the option of putting a mic + interface on an amp is always there but I live in a noisy area. Any advise would be welcome. Thanks!
What would you do if all your source files are 44.1k? (I mostly sample from Apple music and that is what people release) Do I make a general rule of thumb to upsample in RX to 48k for processing or just keep my sessions at 44.1?
Dear Streaky, would you please care to elaborate just a tiny bit on the Pro Tools conversion? I've been told by a rather trustworthy person it was actualy good, or pretty decent at least. I understand you are sure to have used the tweakhead/slowest mode. (?)
What do most or all platforms render down the audio to, without messing with the audio? Is it 44.1 or 48 khz at 16 bit, or 44.1 or 48 khz at 24 bit? This is really confusing? I am going to start doing voice overs with my videos and music alike, what is the highest khz and sample rate to record in, so that I won't have to change it or mess with it later on? Thanks...SBN RESONATE
I’m not an authority on sample rates, but recently, within reaper, I took a session that was recorded at 96k, re-samples all of the tracks within Reaper to 48k, rendered the track to a stereo 48k file. The end result had a stark difference in the upper midrange from the same session bounced from the original 96k resolution tracks, down to a stereo 48k file. Maybe try this out for yourself some time and see what you think!
Hi Steaky can you do a "how to do a sample rate conversion" I have apogee PSX100 that can do multi sample rates but don't know that works. I also Big Ben clock. Big Ben is clocking my apollo 8, Rosetta 800, and the PSX100. I dump the PSX 100 back into wave lab or the session. Do I need a different computer to capture? Thanks. Love your show.
Yes you need 2 computers. One computer will play the original source file and will feed the next interface in the next computer at your desired sample rate. 2 different computers running 2 different sample rates.
Of the millions of comparison videos I’ve seen, I don’t think I’ve seen one comparing sample rate conversion software. Not sure I would know what to listen out for either. I’ve often converted in pro tools without a second thought, so I’d be interested to know what I should be listening for when I get round to doing a comparison for my self?
Hey Streaky, my workflow is daw>analog summing box>daw...my projects are at 48khz but i want to change sample rate at the end to 44.1khz for the client, when coming back into the daw i have a print track set ready to record my project once i am done mixing. But what is the best way to export my file without using the built in bounce within my daw? i am using logic pro x 10.3
agree 100% also the Plugins that offer oversampling can oversample even if you work in 44.1 so i dont think you need to take the project to 96 in order to use their oversampling. Its just lost CPU
48 is good but isn't always the best choice. 96 and 196 do matter if you modify the audio (pitch shifting/audio stretching). For film, 196 might be very important as they may want to mess with the audio in post-production. It only makes a difference if it was recorded in a higher sample rate, though.
@@Streaky_com that's confusing. The comment was about "anything above 48 is nonsense" and you replied that you agree that higher rates are best for HD.
@@OrangeMicMusic Streaky said bit depth, not sample rate. The video was about sample rate, which is the amount of samples per second. The bit depth is the resolution of each sample, like 16 bit or 24 bit.
But what about when just producing music ITB using samples, vst instruments and stuff.. majority of samples are 44.1 so no benefit using 96 when producing ( better plugin sound like you suggested) or just stick to 44, because samples are 44? Thank you
I'm a composer/producer I always work in 48Khz/24bit because my work is orchestral, trailer, TV/Film. I think if you're mixing for streaming services or general pop/EDM then the usual is 44.1Khz/24bit.. so it depends on over all genre/medium.
@@zxy7529 The only thing >48 kHz will accomplish is killing the cpu-cycles. Double the load without any benefits! Plus you have to render to a lower samplerate in the end.
@@Magnus_Loov some plugins do wierd things at nyquist. If you up the nyquist to 40khz (96khz samplerate) than artifacts are not heard. As it is there's not that much difference, but when processing its beneficial to do 96.
In your other video about mastering I understood your advice was to keep at 48Khz 32bit. Is that correct? 6:25 ruclips.net/video/a5tjf0P1ln0/видео.html
I'll go watch ur referenced link but what he's saying is to use the highest setting while mastering but export to 44.1khz for music (iTunes, spotify etc) and 48khz if the music is going in a video.
just a clarification: bit rate is bits per sec (CD quality 16bit/44.1k is 1,441kbps). Bit *depth* (or word length) is bits per sample (16, 24). Most aggregators will accept 16 bit 44.1k. Whereas 24 bit is better suited for lossy encoding, in my experience.
I love how most people record their stuff with shitty mic, shitty interface, shitty performance, shitty somg, and then wants to bounce everything in 96. 🤡
actually, there's still a bandwidth problem, massively. No streaming is real hi-res, even for services like Tidal. Or I should say that they are "hi-res", but never loseless All streaming services will slim down their file sizes, whether it's audible/visible or not. For example, RUclips will always trim down your resolution, both audio and video, doesn't matter if you're using the highest possible setting. For example a 4k video downloaded from RUclips, will always be significantly smaller to the source file that was uploaded, with very minimal or even indistinguishable difference between the two. Also another reason why we're not ditching lower sample rates is lots of electronic devices are still running on "low res" settings. For example, windows default your audio settings to 48kHz. Try switching to 96k and running any slightly demanding games/apps on a typical consumer laptop. It'll most likely chug out big time. So making 96kHz as the minimum resolution, for instance, will mean that everything consumer electronics have to go up in quality, big time. It's kinda like the 8k debate, why multiply your processing load for really very little perceived quality increase. However...I'd be the first one to agree big time if we're ditching 44.1k and using 48k, 96k & all multiples of them. For the sole reason that video requires 48kHz, and 44.1k is a non-rounded figure. So if everyone uses 48k, up & down sampling will have less artifacts in general. But then you have traditions to deal with, for example the broadcast industry has been asking for 44.1k 16bit for ages (at least locally here) and that's hard to change overnight. Also loops/sample banks for DAW's are normally delivered to the users in 44.1k to reduce installation sizes. That's as broad a view as I can put out. Hope this helps. Sorry if it's hella long
@Chad Gillihan really ?? I'm very sceptical of that one. Just seems over kill especially considering today's younger audience don't give a monkeys , music is more throw away than ever unfortunately. So your saying even though you end up at 44.1 you can still hear greater depth in the mix because it was originally recorded mixed ect at 96k ..mmmmmm
@Chad Gillihan That's totally wrong! We shouldn't really talk about "higher resolution" here, because it's not resolution so much as extension. It's not like computer display screens, where 4k is good but it's not as close to perfect as 8k, and 8k will one day be outdone by a yet better resolution, and every step closes the gap to real vision. 44.1kHz is as good as it gets for recording 0-22050Hz, and a 192kHz file is absolutely no better at capturing frequencies in the same range: the data for the frequencies up to 22050Hz are the same in both files (if anyone here doesn't believe me, go into your high sample rate packs and find a sample recorded at 96kHz, 192kHz or whatever you like, make a copy converted to 44.1kHz and invert it against the original: I promise you it's going to null*). All that extra data is just frequencies you can't truly hear but might be able to feel. And that's even questionable, since it's pretty unlikely you're even feeling frequencies above 24kHz, and you might be getting some intermodulation distortion. * I should point out -- if there's dense high freq info going through 22050Hz into the extended range, you won't get a null in that top area because the slope of the upper filter will affect the 44.1 version, making it different to the original. Not worse quality though, just 0.1-3dBs or so change, which is why a 44.1 can sound duller than a 48 or above original It's even questionable if working at a high sample rate has any negative impact on quality with Oversampling, wrt to whether or not internal conversions are detrimental to the signal. I've never found anyone who's been able to prove to me that a sound is better or worse as a result of going into an oversampling plugin at any particular sample rate, and besides, that's all mysterious anyway since you don't know what quality rates are being used in the oversampling. So, on the basis that there's an argument for saying that a certain amount of frequencies above 22050Hz may be "feel-able", and because TV/movies/videogames require me to use 48kHz in the end anyway, I use 48kHz throughout.
Oh shut tf up, you clown. You don't understand the basics of what's being talked about here and you're completely out of your depth. The reason why not to use 44.1 throughout (and why are you putting that word in scare-quotes? Is language beyond you too?) is because 0-22050Hz frames the absolute limits of what a human being with perfect human hearing can identify as a pitch, i.e. if they hear a precise sine at that pitch, or a group of sines around that pitch, can the listener reliably prove they heard a sound being played? BUT, just like we "feel" bass in under-20Hz ranges, we seem to feel higher then 20kHz frequencies. Perhaps their presence reinforces lower frequencies which are within the audible range a tiny bit. That's plausible, but only up to a point. You, like so many other people, have misunderstood those stair-stepped diagrams where someone lays a nasty-looking up-right-down-right etc path over a smooth waveform. You don't like the way that squarish line looks, which is why you start talking about resolution. You think that high sample rate audio will mean more slices and therefore a jagged line that's closer to the curved line, like a high resolution graphic has more pixels and looks more like a true curve. And you're wrong, because that's not how this works. One, the data points in 44.1 are as numerous as they need to be and can't be better in any higher audio format (just 44.1 can't capture the top frequencies of 48, 88.2 etc). Two, although the data points look like they can't be good enough to produce the smooth curve accurately, they actually do so perfectly and I can prove that they do by showing you any 192kHz file you like nulling when it's copied to a 44.1kHz form and inverted against itself. If there is literally any difference between them, they would not null. Oversampling is a different matter-plugins that use oversampling work by temporarily employing a higher sample rate, but they do so for a very specific job (preventing foldback aliasing) which is not the same as running, say, 192kHz all over your project. I know we're accustomed to chuckling at the people who decided standards in the past ("lololol they actually thought this was the best we'd ever need!") but in the case of digital recording the sample rate really is as good as it gets within whatever frequency range it reproduces. And the reason people use higher levels is almost always because they don't understand the maths here and they incorrectly believe that more range must mean more quality. If you're a sound designer who radically alters the dimensions of their recordings, fine: maybe you'll pitch something way down from a super-high sample rate and change a sound that only bats could hear into one that's useful to humans, but aside from that, all that extra data is a waste that's bogging down your computer for no good reason and thus inhibiting how good your music can be. And intermodulation distortion might even introduce some minor issues that you're better off without (btw, *if* your speakers even go up above 21kHz, and that's a big if. If you've paid 5-6 thousand a pair, they *might* go up to ~30kHz which is much less than the 48000Hz max you'll get out of 96kHz audio. Either way, these extra frequencies are largely being thrown out).
So happy with 24/48 never go more than that...
I use Pyramix and Reaper (and/or our Fairlight or LAVRY mastering converters OTB) for sample rate conversion. Love what Pyramix and Reaper have done.
There's a free SRC software called R8Brain by Voxengo for those on a budget.
And it's the next best thing compared to RX
There are plenty of bad sample rates converters that unbalance the sound when down sampling. So imagine the conversion algorithms of the streaming platforms, there must be some real bad ones out there too. It is better, to avoid all these problems, to always upload music using a sample rate of 44.1k to any streaming platform. So as explained in the video by Streaky, a dedicated specialized sample rate conversion tool is recommended, because most of the ones provided with DAWS are crap and unbalance your sound when performing this down sampling conversion. Also, think about this as an engineer when delivering music to an artist: the down sampling conversions that the artist may do with the music that you send to him (for example, when he makes a video clip for RUclips and inserts the sound track into it) may alter the good sound that you made. Make sure to recommend to all artists to only distribute and use the 44.1kHz version and make sure that YOU make it, not them. Otherwise you may see your name on a production that does not sound like the one you made. Best Regards from your friend François in Montreal, Canada! 🎼🃏🎶🎵🙏😁👋🌠
Never had a single issue with uploading at 196... if anything my tracks uploaded to streaming platforms at 196have higher perceived loudness
@@queenpurple8433 A vast majority of high-resolution music files have a sampling rate of 96kHz or 192kHz, both of which are appreciably higher than the 44.1kHz cap on other digital files. High-resolution streaming services allow you to stream music at 96kHz or 192kHz if you have the right gear.. so it may be a good option as well to proceed with your method if you want these platforms to play in your high-res sample rate.
- Apple Music, Spotify, TIDAL and other online stores/streaming services - 16-bit/44.1k WAV files
- Apple Digital Masters - 24-bit/96k, 88.2k, 48k, or 44.1k sample rate WAV files
- Bandcamp and SoundCloud - 24-bit WAV files (sample rates above 44.1k)
- Vinyl - 24-bit WAV Files (sample rates higher than 44.1k if available)
- Cassette - 16-bit WAV Files (sample rates higher than 44.1k in some cases)
- RUclips - 24-bit/48k WAV files or 24-bit/44.1k
Be mindful of the delivery specs for RUclips (or whatever video platform you are using) when the video editor makes their final export. Avoid using mediocre sample rate conversion algorithms that are often found in video editing software, or basic DAWs like Pro Tools, Logic etc. iZotope RX has great sample rate conversion, and most mastering engineers have “mastering grade” sample rate conversion software which can produce better results. I personally use SOX to make the sample rates conversions and my sound keeps a 100% perceived fidelity when making these conversions and comparing both with high-definition headphones
Because your audio for video is likely to encounter some sample rate conversions and data-compression, many recommend leaving at least a full dB of peak-headroom to avoid any clipping due to the conversion processes.
@@themagicianofsound so it's better to downsampling the file before sending to streaming platforms? I would like to know more about this, in my experience I've always upload at 48khz and I've never encountered problems btw
@@AftertuneMusic If you are happy with the results, I would say that you can keep going with the method that you are using. I am more concerned when I give music to artists (especially non-professionals) who then upload on various platforms and also make their own video clip or their own lyrics videos. They use various tools and various conversion methods without knowing what they are doing. It already happened a few times to me that I was hearing my master on their RUclips channel and that the EQ balance was not exactly the same as the master that I delivered (usually the most perceptible symptoms are lighter bass between 63 and 160Hz and higher mids between 2k and 3.5k). After making extensive research and testing, I was able to reproduce the problem myself and this EQ unbalance was happening when down sampling or up sample the sample rate with various DAWs including mine. I now use SOX to perform sample rates conversions and not my DAW. This allows me to keep a perfect sound when performing these conversions. One of the music producers I work with was having the same problem with another DAW when performing sample rate conversions and never noticed the problem until I tell them "Hey wait, your 44,100Hz and 48,000Hz versions have slight differences in the EQ balance that I can perceive". So I prefer now to give to artists a 44,100Hz 24 bits version that I create myself and I specify in the filename "for streaming platforms and RUclips" and another version in 48,000Hz 16 bits and I indicate in the file name "for SoundCloud". So I am sure that the artist or the upload process will not mess up the EQ balance because I am very sensitive to this. But if you are uploading your own music and you feel that your sound is 100% accurate when your music is uploaded, then there is no reason to change your process in my opinion.
@@themagicianofsound thanks a lot for the info, it's always nice to have more opinion in this tecnical issue. I personally now produce and mix at 48khz then i mastered in another session at 96khz and then uploading, but now I want to check more about balance shift.
thanks for taking your time on Saturday and make this video....cheers
I learned that it sounds ok if you divide or multiply by 2 like 88.2 to 44.1 because it's just half or double the samples, but you run into issues if you have to convert from 48 to 44 for instance. Why didn't you say anything about that? Isn't this a big issue anymore with a good converter? It's hard to imagine that if I do the math. I would like to know because this is the reason for music cd's and severses that I go for 88.2 or 44.1. Thanks
Because it's nonsense. It's not the division that's a challenge when converting sample rates. It's the filtering. If you'd just throw away every other sample when converting from 88.2kHz to 44.1kHz you'd be using the most naive and worst sounding conversion possible, with any signal above 22050Hz folding straight back into its aliased equivalent frequency.
@@michaelanderwald4179 That makes sense. Thank you for this explanation. An engineer that I respect for his excellent work explained the double or half rule. But of course, when you take every other sample you get aliasing.
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44.1 kHz and you are done. There is no reason to even record in anything higher kHz. Why? Several reasons. First off: Did you ever hear about the Nyquist-Shannon sampling theorem? Strongly simplified, it proved, that a sampling rate of twice of the highest frequency you want to record is enough to create a perfect digital audio sample. According to this theorem, you can create a perfect range of 1-22050 hz. With any sort of post processing possibly needed you will get this outcome out of a source material at 48kHz of sampling rate. Also, on most audio interfaces, the recording gets worse the higher of a sampling rate you’re trying to record. Even on most prosumer grade audio interfaces or recording devices, 48kHz will sound pristine, while at 96 kHz, recordings will greatly suffer, let alone 192 kHz.
Who would have thought I’d see the day, where a comment does self-destruct? You prove, that nothing is impossible. Thank you for that! 🙂
@DEEZ LIKES I don’t think that argument wins in comparison to the original comment. Especially with a phrase like “educate yourself” to end.
Mate, the higher the sample rate the better recording audio. processing is more accurate. Level meters respond more accurate and thus anything that reads level ie. compressor is more accurate. Distortion and Saturation bounce back with much less Aliasing that effects the audio high and low ends. There's a reason best plugins have oversampling options; they run at a higher sample rate the convert back down to session at say 44.1 kHz. And Im not sure what interface you are using but my simple SSL 2 sounds much better at 96khz that 48khz. The reason I use 96khz is because I mostly use Waves and they don't have an oversampling option..
that's a theorem. in practice you need a perfect filter that will reject all aliasing above niquist, very hard to build. hence, increasing the sample rate will move errors/ aliasing into inaudible spectrum . apparently 48khz willdo the trick, yet 60khz is perfect yet nowhere to be found in common recorders. so, 96khz it is!
Over 48 you will have oversampling issues with most plugins. It have being demonstrated already.
Please note that You can’t downsample from 96khz to 44.1khz. They are not compatible ! and will alter the sound… recording for 44.1khz must be made in 88.2khz and For video format in 48khz you should record at 96khz.
@@nicksterj i didn’t say it s not possible. It’s not compatible. I suggest you Do some research on 44.1khz and 48khz . You will altered your source sound going from one to the other. They are not compatible
I have a hard time hearing improvements over 44! maybe because of the kit I use or maybe because I have listened to so much at 44. so much music I loved was made at 44.
Really depends on your DAC. Use a Lynx Hilo at 96k and you will definitely hear a difference!
Your monitors probably cut everything off above 22,050hz.
@@chadm4976 placebo
43.2
For SRC R8Brain is on par with Izotope RX and there is a free version !
I have r8 brain free and pro versions and they both are actually very bad.
@@380stroker have you found a solution that’s better?I’ve used both RX and R8Brain with great success for batch processing: I’ve also used the old school analog way of playing back each tune in real-time and converting sample rate between two converters and separate computers … but that’s not efficient in time management…. Please let me know which other solution has been working for you for batch processing… I’d love to give it a go
@@ZiwaHD For the sake of the client, its always quality over quantity. I don't care how long it takes as long as the end product is optimal. R8brain is horrible. Seriously.
There’s so much saturation and other distortion added to tracks it seems ridiculous to do it at 96khz unless you’re doing classical.
Distortion is the main thing that produces high frequencies. So while I'd have no qualms about doing classical music at 44kHz, I'd really prefer higher sample rates for distorted stuff, especially when I'm actually producing the distortion in the digital domain.
Adding distortion and harmonics is the main reason for using 96Khz in my opinion.
About 88,2 Khz? I use this sampling frequency because, like Bob Katz says, is perfect and gives less errors in 44,1 Khz downsampling .What do you think?
I think Bob a little wacko 😂😂😂 he’s the digital king though 👑
That's actually incorrect (sorry Bob). If you do some research, sine waves are drawn the same, as long as they are not above the nyquist frequency. I find 96k to be a good balance of I/O channel count, cpu power use, and headroom (still should use 4x oversampling for saturation and certain other plugins when possible).
Ah yes, Bob K and his K-system. Look, sample at whatever rate you want. Just make sure at the end you use hardware instead of software to do your last conversion.
Sure it's mentioned already but your thumb nail image says HZ and not KHZ
And yeah it’s KHz.!
That’s what you get for for money on Fiverr 😂😂😂
Can you guys really hear the difference?
Yes
yes
no
100% yes.
Hi Streaky what is the best setting for downsampling from 96 to 44.1 for izotope rx .Thanks for your help
+1
Thanks Streaky, great advice!
Where did all the "k"s go?? 48Hz?
Is there a Sennheiser hd820 review on the horizon? Preferebly compare it to hd800s 🤓
Great content
96 Khz is way overkill for just producing songs in the box.
The only difference you will notice is the more than double load on the CPU, meaning much less plug-ins.
It's a total and unjustifiable waste of CPU-resources. It's gonna end up in 44.1KHz/16 bit anyway and, as said, when needed vst-i:s will upsample (higher samplereate) internally and the DAW already handles things with 32 or even 64 bits internally.
So everything that is important gets the "higher treatment" already WITHOUT the extra CPU-load overall for every plug-in.
Especially if only working with non-acoustic stuff.
Mastering acoustic music may be a totally different thing that I don't know anything about though. But checking about A/B blindtests people can't hear a difference when you go above 44.1/16 bits. There's a reason why it was chosen in the first place for the CD-standard.
You mentioned one VERY important point - I mainly do live recording of bands and orchestras and ONLY record at 96k to capture a good hi resolution signal. If you are working in the box on EDM etc with samples that were created at 44.1 or have come from mp3 then sure - there is no point in working above 44.1k
If I'm not mistaken, for multitrack recording of a singer or instrumentalist a higher sample rate helps to achieve lower latency.
@@hanslutz6420 My soundcard will get 2 ms latency at 48 samples @44.1 kHz(1 ms in and 1 ms out), which is more than enough for anything. I have it set at 256 samples buffer right now...
@@hanslutz6420 This is true
I have to agree with all of that
Hi, one slightly basic home recordist question. Would you advise I get a multi effects guitar pedal that outputs at 44.1 kHz or one that outputs at 48 kHz. Like you mentioned, audio typically requires 44.1 khz and video requires 48 khz. Since I intend to use the unit on both types of projects, would upsampling the guitar parts be better, or downsampling? As work flow goes I’d need to record it in the device’s native sampling rate, then convert it into the project’s sample rate, and then import it into the project - would there be any better way to do this. Of course, the option of putting a mic + interface on an amp is always there but I live in a noisy area. Any advise would be welcome. Thanks!
@MF Nickster Yes, thanks, that does seem like the most sensible option as I can let the audio interface decide the sample rate :)
maaaaaannnnnnnnnnnnn people! what format you still have to export out?????
Haha, wut?
What would you do if all your source files are 44.1k? (I mostly sample from Apple music and that is what people release)
Do I make a general rule of thumb to upsample in RX to 48k for processing or just keep my sessions at 44.1?
Dear Streaky, would you please care to elaborate just a tiny bit on the Pro Tools conversion? I've been told by a rather trustworthy person it was actualy good, or pretty decent at least. I understand you are sure to have used the tweakhead/slowest mode. (?)
why does the thumbnail have 44.1hz etc. is it not Khz??? :-)
Typo
Would you mind changing your thumbnail to show kHz instead of Hz, please? :)
If you cut everything below 20 kHz, many people will hear almost nothing 😅. I believe 20 Hz is correct regarding subsonic frequencies...
Thanks
What do most or all platforms render down the audio to, without messing with the audio? Is it 44.1 or 48 khz at 16 bit, or 44.1 or 48 khz at 24 bit? This is really confusing?
I am going to start doing voice overs with my videos and music alike, what is the highest khz and sample rate to record in, so that I won't have to change it or mess with it later on? Thanks...SBN RESONATE
I’m new to self-recording but quite tek savvy. How are the resamplers in Reaper and Amadeus Pro? Do I need to invest in a dedicated sampler programme?
I’m not an authority on sample rates, but recently, within reaper, I took a session that was recorded at 96k, re-samples all of the tracks within Reaper to 48k, rendered the track to a stereo 48k file. The end result had a stark difference in the upper midrange from the same session bounced from the original 96k resolution tracks, down to a stereo 48k file. Maybe try this out for yourself some time and see what you think!
A great majority of pros do realtime hardware sample rate conversion. It just sounds better.
If you want to hear good sample rate conversion, shoot me a message.
Hi Steaky can you do a "how to do a sample rate conversion" I have apogee PSX100 that can do multi sample rates but don't know that works. I also Big Ben clock. Big Ben is clocking my apollo 8, Rosetta 800, and the PSX100. I dump the PSX 100 back into wave lab or the session. Do I need a different computer to capture? Thanks. Love your show.
Yes you need 2 computers. One computer will play the original source file and will feed the next interface in the next computer at your desired sample rate. 2 different computers running 2 different sample rates.
Love your channel. The link for the newsletter doesn't work. It gives a 404 error.
Of the millions of comparison videos I’ve seen, I don’t think I’ve seen one comparing sample rate conversion software. Not sure I would know what to listen out for either. I’ve often converted in pro tools without a second thought, so I’d be interested to know what I should be listening for when I get round to doing a comparison for my self?
Stereo image and clarity.
But here in India the cinema exhibition companies take audio at 48/16 sample rate only.
48khz is the worlds video standard sampling rate.
I read some of the comments down here and man o man people are really lost on this topic lol
And do you convert to 32bit float as well coming in?
No, always use integer. Your DAW works in floating point to begin with.
@@380stroker I use Wavelab 10 now ie in audio montage, which defaults to 64bit float regardless. problem solved
Good video thank no bs
Khz not hz.
44.1khz is the best frequency
That makes no sense, lol. Do you mean it's the best sampling rate in your opinion?
@@380stroker r/wooosh
@@380strokeryes that’s what he means. It’s a frequency of sample rate.
cla says its better to mix in 44.1 or 48 because if you go higher, when you downsample you lose high frecuencies, with izotope rx that still happends?
CLA mixes from digital tape machine
Which sample rate should I choose to streaming with obs ? 44.1k or 48 k ?
44
any reason to NOT use 192?
Hey Streaky, my workflow is daw>analog summing box>daw...my projects are at 48khz but i want to change sample rate at the end to 44.1khz for the client, when coming back into the daw i have a print track set ready to record my project once i am done mixing. But what is the best way to export my file without using the built in bounce within my daw? i am using logic pro x 10.3
get an external print box. sold state recorder ect
Send it to me. I'll show you what's up.
48 is scientifically proven to be where it’s at.Anything above that makes no sense. Hire rates are marketing mambo jambo.
I agree I always find bit depth makes for more HD sound
agree 100% also the Plugins that offer oversampling can oversample even if you work in 44.1 so i dont think you need to take the project to 96 in order to use their oversampling. Its just lost CPU
48 is good but isn't always the best choice.
96 and 196 do matter if you modify the audio (pitch shifting/audio stretching). For film, 196 might be very important as they may want to mess with the audio in post-production. It only makes a difference if it was recorded in a higher sample rate, though.
@@Streaky_com that's confusing. The comment was about "anything above 48 is nonsense" and you replied that you agree that higher rates are best for HD.
@@OrangeMicMusic Streaky said bit depth, not sample rate. The video was about sample rate, which is the amount of samples per second. The bit depth is the resolution of each sample, like 16 bit or 24 bit.
But what about when just producing music ITB using samples, vst instruments and stuff.. majority of samples are 44.1 so no benefit using 96 when producing ( better plugin sound like you suggested) or just stick to 44, because samples are 44? Thank you
I'm a composer/producer I always work in 48Khz/24bit because my work is orchestral, trailer, TV/Film. I think if you're mixing for streaming services or general pop/EDM then the usual is 44.1Khz/24bit.. so it depends on over all genre/medium.
@@zxy7529 The only thing >48 kHz will accomplish is killing the cpu-cycles. Double the load without any benefits! Plus you have to render to a lower samplerate in the end.
@@Magnus_Loov some plugins do wierd things at nyquist. If you up the nyquist to 40khz (96khz samplerate) than artifacts are not heard. As it is there's not that much difference, but when processing its beneficial to do 96.
Does changing the sample rate change the bpm?
If you don't convert it : yes
60khz is the ideal sample rate. Fight me if you disagree.
I trust Dan Lavry too
The highest possible
88.2
In your other video about mastering I understood your advice was to keep at 48Khz 32bit. Is that correct? 6:25 ruclips.net/video/a5tjf0P1ln0/видео.html
I'll go watch ur referenced link but what he's saying is to use the highest setting while mastering but export to 44.1khz for music (iTunes, spotify etc) and 48khz if the music is going in a video.
Do not mistake 32 bit integer for 32 bit floating point. Having said that, don't mixdown to 32 bit floating point.
what about bitrates? what are you most commonly distributing? 16 or 24?
just a clarification: bit rate is bits per sec (CD quality 16bit/44.1k is 1,441kbps).
Bit *depth* (or word length) is bits per sample (16, 24).
Most aggregators will accept 16 bit 44.1k. Whereas 24 bit is better suited for lossy encoding, in my experience.
Video audio is most often 48 so I would say 48
I love how most people record their stuff with shitty mic, shitty interface, shitty performance, shitty somg, and then wants to bounce everything in 96. 🤡
Im guessing 96khz
So just keep it at 96
Come on mate, fix the typo in your thumbnail. You literally got wrong the one measurement unit the entire video is predicated on.
As hi-res streaming isn't really a bandwidth problem anymore, why not aim for 96 in all media and leave 44.1 as a "low-res" option?
actually, there's still a bandwidth problem, massively. No streaming is real hi-res, even for services like Tidal. Or I should say that they are "hi-res", but never loseless
All streaming services will slim down their file sizes, whether it's audible/visible or not. For example, RUclips will always trim down your resolution, both audio and video, doesn't matter if you're using the highest possible setting. For example a 4k video downloaded from RUclips, will always be significantly smaller to the source file that was uploaded, with very minimal or even indistinguishable difference between the two.
Also another reason why we're not ditching lower sample rates is lots of electronic devices are still running on "low res" settings.
For example, windows default your audio settings to 48kHz. Try switching to 96k and running any slightly demanding games/apps on a typical consumer laptop. It'll most likely chug out big time.
So making 96kHz as the minimum resolution, for instance, will mean that everything consumer electronics have to go up in quality, big time.
It's kinda like the 8k debate, why multiply your processing load for really very little perceived quality increase.
However...I'd be the first one to agree big time if we're ditching 44.1k and using 48k, 96k & all multiples of them. For the sole reason that video requires 48kHz, and 44.1k is a non-rounded figure. So if everyone uses 48k, up & down sampling will have less artifacts in general. But then you have traditions to deal with, for example the broadcast industry has been asking for 44.1k 16bit for ages (at least locally here) and that's hard to change overnight.
Also loops/sample banks for DAW's are normally delivered to the users in 44.1k to reduce installation sizes.
That's as broad a view as I can put out. Hope this helps. Sorry if it's hella long
everything is going to end up on spotify, itunes, youtube ect... what is the point of 96k??? just dont get it !!
@Chad Gillihan really ?? I'm very sceptical of that one. Just seems over kill especially considering today's younger audience don't give a monkeys , music is more throw away than ever unfortunately. So your saying even though you end up at 44.1 you can still hear greater depth in the mix because it was originally recorded mixed ect at 96k ..mmmmmm
@Chad Gillihan fair enough man..always good to discuss these things .. respect
@Chad Gillihan That's totally wrong! We shouldn't really talk about "higher resolution" here, because it's not resolution so much as extension. It's not like computer display screens, where 4k is good but it's not as close to perfect as 8k, and 8k will one day be outdone by a yet better resolution, and every step closes the gap to real vision. 44.1kHz is as good as it gets for recording 0-22050Hz, and a 192kHz file is absolutely no better at capturing frequencies in the same range: the data for the frequencies up to 22050Hz are the same in both files (if anyone here doesn't believe me, go into your high sample rate packs and find a sample recorded at 96kHz, 192kHz or whatever you like, make a copy converted to 44.1kHz and invert it against the original: I promise you it's going to null*). All that extra data is just frequencies you can't truly hear but might be able to feel. And that's even questionable, since it's pretty unlikely you're even feeling frequencies above 24kHz, and you might be getting some intermodulation distortion.
* I should point out -- if there's dense high freq info going through 22050Hz into the extended range, you won't get a null in that top area because the slope of the upper filter will affect the 44.1 version, making it different to the original. Not worse quality though, just 0.1-3dBs or so change, which is why a 44.1 can sound duller than a 48 or above original
It's even questionable if working at a high sample rate has any negative impact on quality with Oversampling, wrt to whether or not internal conversions are detrimental to the signal. I've never found anyone who's been able to prove to me that a sound is better or worse as a result of going into an oversampling plugin at any particular sample rate, and besides, that's all mysterious anyway since you don't know what quality rates are being used in the oversampling.
So, on the basis that there's an argument for saying that a certain amount of frequencies above 22050Hz may be "feel-able", and because TV/movies/videogames require me to use 48kHz in the end anyway, I use 48kHz throughout.
Oh shut tf up, you clown. You don't understand the basics of what's being talked about here and you're completely out of your depth. The reason why not to use 44.1 throughout (and why are you putting that word in scare-quotes? Is language beyond you too?) is because 0-22050Hz frames the absolute limits of what a human being with perfect human hearing can identify as a pitch, i.e. if they hear a precise sine at that pitch, or a group of sines around that pitch, can the listener reliably prove they heard a sound being played? BUT, just like we "feel" bass in under-20Hz ranges, we seem to feel higher then 20kHz frequencies. Perhaps their presence reinforces lower frequencies which are within the audible range a tiny bit. That's plausible, but only up to a point.
You, like so many other people, have misunderstood those stair-stepped diagrams where someone lays a nasty-looking up-right-down-right etc path over a smooth waveform. You don't like the way that squarish line looks, which is why you start talking about resolution. You think that high sample rate audio will mean more slices and therefore a jagged line that's closer to the curved line, like a high resolution graphic has more pixels and looks more like a true curve. And you're wrong, because that's not how this works. One, the data points in 44.1 are as numerous as they need to be and can't be better in any higher audio format (just 44.1 can't capture the top frequencies of 48, 88.2 etc). Two, although the data points look like they can't be good enough to produce the smooth curve accurately, they actually do so perfectly and I can prove that they do by showing you any 192kHz file you like nulling when it's copied to a 44.1kHz form and inverted against itself. If there is literally any difference between them, they would not null. Oversampling is a different matter-plugins that use oversampling work by temporarily employing a higher sample rate, but they do so for a very specific job (preventing foldback aliasing) which is not the same as running, say, 192kHz all over your project.
I know we're accustomed to chuckling at the people who decided standards in the past ("lololol they actually thought this was the best we'd ever need!") but in the case of digital recording the sample rate really is as good as it gets within whatever frequency range it reproduces. And the reason people use higher levels is almost always because they don't understand the maths here and they incorrectly believe that more range must mean more quality. If you're a sound designer who radically alters the dimensions of their recordings, fine: maybe you'll pitch something way down from a super-high sample rate and change a sound that only bats could hear into one that's useful to humans, but aside from that, all that extra data is a waste that's bogging down your computer for no good reason and thus inhibiting how good your music can be. And intermodulation distortion might even introduce some minor issues that you're better off without (btw, *if* your speakers even go up above 21kHz, and that's a big if. If you've paid 5-6 thousand a pair, they *might* go up to ~30kHz which is much less than the 48000Hz max you'll get out of 96kHz audio. Either way, these extra frequencies are largely being thrown out).
@@joechapman8208 Dude, you are so wrong and sound uneducated on the subject.
Why not just record at 44.1 to begin with rather than throw away all that extra information every time to get to your final product?
This is generally mistaken advice.
I dont use pro tools but cubase, a lot better sounding than pro tools.
no daw sound better than other one