6 ms·

How problematic is resampling audio from 44.1 to 48 kHz?

51 points by brewmarche 9mo ago

ZeroConcerns 9mo ago
> it's probably worth avoiding the resampling of 44.1 to 48 kHz Ehhm, yeah, duh? You don't resample unless there is a clear need, and even then you don't upsample and only downsample, and you tell anyone that tries to convince you otherwise to go away and find the original (analog) source, so you can do a proper transfer.
- mort96 9mo ago
  If only it was that simple T_T I'm working on a game. My game stores audio files as 44.1kHz .ogg files. If my game is the only thing playing audio, then great, the system sound mixer can configure the DAC to work in 44.1kHz mode. But if other software is trying to play 48kHz sound files at the same time? Either my game has to resample from 44.1kHz to 48kHz before sending it to the system, or the system sound mixer needs to resample it to 48kHz, or the system sound mixer needs to resample the other software from 48kHz to 44.1kHz. Unless I'm missing something?
  - mort96 9mo ago
    And actually, why do we have both 48kHz and 44.1kHz anyway? If all "consumer grade high quality audio" was in 44.1kHz (or 48kHz) we probably could've avoided resampling in almost all circumstances other than professional audio contexts (or for already low quality audio like 8kHz files). What benefit do we get out of having both 44.1 and 48 that outweighs all the resampling it causes?
    - deleted 9mo ago
      [deleted]
    - zamadatix 9mo ago
      As far as I understood, both rates ultimately come from trying to map to video standards of the time. 44.1 kHz mapped great to reusing analog tape of the time, 48 kHz mapped better to digital clocking and integer multiples of video standards while also having a slightly wider margin on oversampling the high frequency. 44.1 kHz never really went away because CDs continued using it, allowing them to take any existing 44.1 kHz content as well as to fit slightly more audio per disc. At the end of the day, the resampling between the two doesn't really matter and is more of a minor inconvenience than anything. There are also lots of other sampling rates which were in use for other things too.
    - CharlesW 9mo ago
      > And actually, why do we have both 48kHz and 44.1kHz anyway? Those two examples emerged independently, like rail standards or any number of other standards one can cite. That's really just the top of the rabbit-hole, since there are 8-20 "standard" audio sample rates, depending how how you count. This isn't really a drawback, and it does provide flexibility when making tradeoffs for low bitrates (e.g. 8 kHz narrowband voice is fine for most use cases) and for other authoring/editing vs. distribution choices.
      - Joeboy 9mo ago
        > This isn't really a drawback But, that's only true because people freely resample between them all the time and nobody knows or cares about it.
      - pixelpoet 9mo ago
        The nice thing about standards is, there are so many from which to choose! :)
    - sneak 9mo ago
      CDs used 44.1, DAT and DVDs used 48. That’s it. Most stuff on the internet ripped from CD is 44.1. 48 is getting more common. We’re like smack in the middle of the 75 year transition period to 48kHz. For new projects, I use 48, because my mics are 32bit (float!)/48kHz.
    - brudgers 9mo ago
      Perhaps the benefit we get is access to existing recordings? 44.1khz exists because it was the lowest technically practical speed and was an optimization for processing speed and storage space. 48khz exists because it syncs with video easily — I’ve also heard it allows for more tolerance in the anti-aliasing filter.
      - cormorant 9mo ago
        > 48khz exists because it syncs with video easily I guess meaning 24fps video? Because 44100 is already a multiple of 25, 30, 50, and 60.
        adgjlsfhk1 9mo ago
        48k also supports 24, 48, 120, and 240 which are all nice to haves.
        brudgers 9mo ago
        24fps is where the money is because 24fps is the standard for film.
    - PunchyHamster 9mo ago
      technically we could use 40kHz and just upsample, the extra frequency over 40kHz is basically leeway to make analog part possible/cheap, but it is not technically needed in the signal the first CD player didn't had compute power to upsample perfectly but modern devices certainly do.
      - mort96 9mo ago
        AFAIU, 40kHz exactly wouldn't really work, if your goal is to represent 0Hz-20kHz: in order to avoid aliasing, you need a low pass filter to remove all frequency content above half your sample rate, and no filter is infinitely hard (and you generally want to give the filter a decent range of frequencies to work with). If you want to start your low pass filter at 20kHz, you want it to end (i.e reach practically -∞dB) at a few kHz above 20kHz. If you used a sample rate of exactly 40kHz, you would need your low pass filter to reach -∞dB at 20kHz, meaning it'd have to start somewhere in the audible region. Though this is just my understanding. Maybe I'm wrong.
    - rerdavies 9mo ago
      > Why do we have both 48kHz and 44.1kHz anyway Because of greed. Early audio manufacturers (SONY notably) used 48kHz for profession-grade audio equipment, that would be used in studios or TV stations, and degraded 44.1khz audio for consumer devices. Typically you would pay an order of magnitude more for the 48kHz version of the hardware. 48khz is better for creating and mixing audio. You cannot practically mix audio at 44.1khz without doing very slight damage to audible high frequencies. But enough to make a difference. If you were creating for consumer devices, you would mix at 48Khz, and then downsample to 44.1khz during final mastering, since conversion from 48kHz to 44.1kHz can be done theoretically (and practically) perfectly. (Opinions of the OP notwithstanding). I think it's safe to say that the 44.1kHz sampling rate was maliciously selected specifically because it is just low enough that perfect playback is still possible, but perfect mixing is practically not possible. And obviously maliciously chosen to be a rate with no convenient greatest common denominator with 48Khz, which would have allowed easy and cheap perfect realtime resampling. Had Sony chose 44.0kHz, it would be trivially easy to do sample rate conversion to 48Khz in realtime even with primitive hardware available in the late 1970s. That extra .1kHz is transparently obvious malice and greed in plain sight. Presumably SONY would sell you the software or hardware to perform perfect non-realtime conversion of audio from 48khz to 44.1khz for a few tens of thousands of dollars. Not remotely subtle how greedy all of this was. There has been no serious reason to use 44.1kHz instead of 48kHz for about 50 years, at least from a technology point of view. (And no real reason to EVER use 44.1khz instead of 48kHz other than GREED).
      - brendyn 9mo ago
        Are you able to share evidence for this?
        rerdavies 9mo ago
        What would you consider evidence? Emails between standards committee members agreeing to collude in order to screw pro-audio customers? The evidence is: why on earth would anyone on a standards committee choose 44.1kHz, instead of 44.0kHz? The answer: 44.1kHz was transparently obviously chosen to make it impossible to perform on-the-fly rate conversions. The mathematics of polyphase rate converters was perfectly well understood at the time these standards were created.
  - zamadatix 9mo ago
    That's a clear need IMO, but it'd be slightly better if the game could have 48 kHz audo files and downsampled them to 44.1 kHz playback than the other way around (better to downsample than upsample).
    - adgjlsfhk1 9mo ago
      They're both fine (as long as the source is band limited to 20khz which it should be anyway).
      - adrian_b 9mo ago
        The analog source is never perfectly limited to 20 kHz because very steep filters are expensive and they may also degrade the signal in other ways, because their transient response is not completely constrained by their amplitude-frequency characteristic. This is especially true for older recordings, because for most newer recordings the analog filters are much less steep, but this is compensated by using a much higher sampling frequency than needed for the audio bandwidth, followed by digital filters, where it is much easier to obtain a steep characteristic without distorting the signal. Therefore, normally it is much safer to upsample a 44.1 kHz signal to 48 kHz, than to downsample 48 kHz to 44.1 kHz, because in the latter case the source signal may have components above 22 kHz that have not been filtered enough before sampling (because the higher sampling frequency had allowed the use of cheaper filters) and which will become aliased to audible frequencies after downsampling. Fortunately, you almost always want to upsample 44.1 kHz to 48 kHz, not the reverse, and this should always be safe, even when you do not know how the original analog signal had been processed.
        PunchyHamster 9mo ago
        yeah but you can record it in 96kHz, then resample it perfectly to 44.1 (hell, even just 40) in digital domain, then resample it back to 48kHz before sending to DAC
        adrian_b 9mo ago
  - zipy124 9mo ago
    you're not missing something. You can re-sample them safely as stated by the author. They simply state you should check the re-sampler as: > Although this conversion can be done in such a way as to produce no audible errors, it's hard to be sure it actually is. That is, you should verify the re-sampler you are using or implement yourself in order to be sure it is done correctly, and that with todays hardware it is easily possible.
  - Veliladon 9mo ago
    Getting pristine resampling is insanely expensive and not worth it. If you have a mixer at 48KHz you'll get minor quantization noise but if it's compressed already it's not going to do any more damage than compression already has.
  - ZeroConcerns 9mo ago
    DAC frequency and the audio format requirements for whatever you supply to your platform audio API have literally nothing to do with each other. My reply was from an audio mastering perspective.
  - Joeboy 9mo ago
    > Unless I'm missing something? I suppose the option you're missing is you could try to get pristine captures of your samples at every possible sample rate you need / want to support on the host system.
  - zelphirkalt 9mo ago
    Is this not the job of the operating system or its supporting parts, to deal with audio from various sources? It should not be necessary to inspect the state of the OS your game is running on, to know what kind of audio you can playback. In fact, that could even be considered spying on things you shouldn't. Maybe the OS or its sound system does not abstract that from you and I am wrong about the state of OS in reality, but this seems to me like a pretty big oversight, if true. If I extrapolate from your use-case, then that would mean any application performing any playback of sound, needs to inspect whether something else is running on the system. That seems like a pretty big overreach. As an example, lets say I change frequency in Audacity and press the play button. Does Audacity now go and inspect, whether anything else on my system is making any sound?
    - Joeboy 9mo ago
      > Is this not the job of the operating system or its supporting parts, to deal with audio from various sources I think that's the point? In practice the OS (or its supporting parts) resample audio all the time. It's "under the hood" but the only way to actually avoid it would be to limit all audio files and playback systems to a single rate.
      - zelphirkalt 9mo ago
        I don't understand then, why they need to deal with that when making a game, unless they are not satisfied with the way that the OS resamples under the hood.
        Joeboy 9mo ago
        My reading is not that they're saying it's something they necessarily have deal with themselves, but that it's something they can't practically avoid.
        rerdavies 9mo ago
        But they CAN practically avoid it. lol. Just let the system do it for them.
  - adzm 9mo ago
    You are right; the system sound mixer should handle all resampling unless you explicitly take exclusive control of the audio device. On Windows at least, this means everything generally gets resampled to 48khz. If you are trying to get the lowest latency possible, this can be an obstacle... on the order of single digit milliseconds.
- bob1029 9mo ago
  If 44.1kHz is otherwise sufficient but you have a downstream workflow that is incompatible, there are arguments for doing this. It can be done with no loss in quality. From an information theory perspective, this is like putting a smaller pipe right through the middle of a bigger one. The channel capacity is the only variable that is changing and we are increasing it.
  - anonymars 9mo ago
    This isn't the whole picture because they aren't multiples of each other, so there will be interpolation/jitter For example if you watch a 24fps film on a 60fps screen, in contrast to a 120fps screen
    - mort96 9mo ago
      That's not how audio works. PCM data at some sample rate with infinite precision samples perfectly encodes all the audio data of all frequencies up to half the sample rate. Resampling is a theoretically lossless operation when the frequency content of the audio fits within half the sample rate of both the source and the destination sample rates (which will always be true when resampling to a higher sample rate, FWIW). The issues are that 1) resampling has a performance and latency cost, 2) better resampling has a higher performance and latency cost
- zipy124 9mo ago
  That seems a rather shallow - and probably incorrect - reading of the source. This is an efficiency and trust trade off as noted: > given sufficient computing resources, we can resample 44.1 kHz to 48 kHz perfectly. No loss, no inaccuracies. and then further > Your smartphone probably can resample 44.1 kHz to 48 kHz in such a way that the errors are undetectable even in theory, because they are smaller than the noise floor. Proper audio equipment can certainly do so. That is you don't need the original source to do a proper transfer. The author is simply noting > Although this conversion can be done in such a way as to produce no audible errors, it's hard to be sure it actually is. That is that re-sampling is not a bad idea in this case because it's going to have any sort of error if done properly, it's just that the Author notes you cannot trust any random given re-sampler to do so. Therefore if you do need to resample, you can do so without the analog source, as long as you have a re-sampler you can trust, or do it yourself.
  - sgerenser 9mo ago
    Speaking of a resampler you trust, I’ve had good experience with libsamplerate (http://www.mega-nerd.com/SRC/ http://www.mega-nerd.com/SRC/), which as of 2016 is BSD licensed.
- brudgers 9mo ago
  A very common clear need is incorporating 44.1khz audio sourcesinto video. 48khz is 48khz because 48khz divided by 24fps, 25fps, or 30fps is an integer (and 44.1khz is not). Also, for decades upsampling on ingest and downsampling on egress has been standard practice for DSP because it reduces audible artifacts from truncation and other rounding techniques. Finally, most recorded sound does not have an original analog source because of the access digital recording has created…youtube for example.
AshamedCaptain 9mo ago
Lots of Live/Audigy era Creative sound cards would resample everything to 48kHz, with probably one of the worst quality resamplers available, to the chagrin of all bitperfect fanatics... still probably one of their best selling sound cards. I.e. no one cares.
- Tanoc 9mo ago
  I had a Soundblaster Live! Gold card back in the day, and I would route my record player or stereo through it so I could use a visualizer on my computer. You could hear the digital noise that was introduced on the highhats. And the source for the sound was a late '70s era Realistic system where everything was analogue. I never knew it was because of the soundcard. I'd always just chalked it up to either Windows XP or VLC doing something.
  - AshamedCaptain 9mo ago
    Unlikely to be this issue since this is about resampling of (e.g. from a CD's native) 44.1kHz PCM to 48kHz, and has nothing at all to do when recording since you'd most likely record at 48kHz and play at 48kHz (no HW resampling involved).
everfrustrated 9mo ago
Changing the sample rate of audio only affects the frequency range. All audio signal is _perfectly_ represented in a digital form. I am ashamed to admit this took me a long time to properly understand. For further reading I'd recommend: https://people.xiph.org/~xiphmont/demo/neil-young.html https://people.xiph.org/~xiphmont/demo/neil-young.html https://www.youtube.com/watch?v=cIQ9IXSUzuM https://www.youtube.com/watch?v=cIQ9IXSUzuM
- pxndxx 9mo ago
  Sampling does not lose information below the Nyquist limit, but quantization does introduce errors that can't be fixed. And resampling at a different rate might introduce extra errors, like when you recompress a JPEG.
  - brudgers 9mo ago
    In the audio world, quantization is usually discussed in terms of bit-depth rather than sample rate.
    - mort96 9mo ago
      Yeah, they know and their comment reflects that knowledge. They're saying that if we had infinite bit depth, we could arbitrarily resample anything to anything as long as the sample rate is above the Nyquist frequency; however we don't have an infinite bit depth, we have a finite bit depth (i.e the samples are quantized), which limits the dynamic range (i.e introduces noise). This noise can compound when resampling.
      - adgjlsfhk1 9mo ago
        The key point is that even with finite bit depth (as long as you dither properly), the effect of finite bit depth is easily controlled noise of program chosen spectrum. i.e. as long as your sampling isn't doing anything really dumb, the noise introduced by sampling is well below noise floor.
cozzyd 9mo ago
Naively I would upsample by 4-8 by 0 stuffing and low pass filtering and then interpolating. That can't be that bad, can it?
MontagFTB 9mo ago
> We do [cubic curve fitting] all the time in image processing, and it works very well. It would probably work well for audio as well, although it's not used -- not in the same form, anyway -- in these applications. Is there a reason the solution that "works very well" for images isn't/can't be applied to audio?
- amlib 9mo ago
  AFAIK it introduces harmonic distortion
  - pixelpoet 9mo ago
    I'd love to know more about this, do you perhaps have any refs? Thanks
    - amlib 9mo ago
      Not an expert in this field, just a scrub, so I can't really give you much. There is this website that has painstakingly compares many resampling algorithms from all sorts of software: https://src.infinitewave.ca https://src.infinitewave.ca Try it's mirror if you can't access it: https://megapro17.github.io/src/index.html https://megapro17.github.io/src/index.html The only one that says it is a cubic interpolation is the "Renoise 2.8.0 (cubic)" one, the spectrogram isn't very promising with all sorts of noise, intermodulation and aliasing issues. And, by switching to the 1khz tone spectrum view you can see some harmonics creeping up. When I used to mess with trackers I would sometimes chose different interpolations and bicubic definitely still colored the sound, with sometimes enjoyable results. Obviously you don't want that as a general resampler...
      - butterknife 9mo ago
        Just to note that this site hasn't been updated for a while. Much better, more modern and with automated upload analysis site would be [1] although it is designed for finding the highest fidelity resampler rather than AB comparisons. [1] https://src.hydrogenaudio.org https://src.hydrogenaudio.org
bobbylarrybobby 9mo ago
Where does dither fit into this picture?
- mistrial9 9mo ago
  dither is not compression or re-sampling. Dither is a means to change data so that it appears similar to original data to a human AFAIK source- wrote dithering code for digital images
  - pixelpoet 9mo ago
    Dithering is about trading spatial / time resolution for reduced average quantisation error.
    - mistrial9 9mo ago
      The word dither -- evidently, the term dithering originally referred to a trembling or erratic movement, from the Middle English didderen, meaning "to tremble." further, the halftone technique developed in the 1880s by Georg Meisenbach — breaking images into dots to simulate shades of gray — was called autotype, not dithering. The term dithering was later adopted in digital imaging and computing, particularly in the 1960s, when engineers applied the concept of adding noise to reduce color banding.
- Sesse__ 9mo ago
  You generally want to dither when (before) you quantize, unless you have so much headroom that it doesn't matter. E.g., if you're converting 44.1/16 to 48/16 (which involves quantizing each sample from an intermediate higher-precision result), you probably want to dither, but if you're converting 44.1/24 to 48/24, you probably won't need to care since you don't really care about whether your effective result is 24 or 24.5–25 bits.
- ErroneousBosh 9mo ago
  That's more of a bit depth rather than bit rate thing. I was surprised to find that going from 16 to 8 bits by simply truncating gave really obvious artifacts on certain sounds (a sampled 808 kick for example had a distinct BZZZEEEOOOOWWWW sound, quite prominent), and even really simple triangular noise dithering made it go away. It did mean the playback was more noisy but it was less obvious. As an aside, G.711 codecs use a kind of log scale with only four bits of signal but small signal values use much smaller bits.
pixelpoet 9mo ago
I'm not sure I understand the "just generate it" perspective. If you want to generate a much higher sampling rate signal that has a common multiple of your input and output sampling rate, "just generating it" is going to involve some kind of interpolation, no? Because you're trying to make data that isn't there. If you want to change the number of slices of pizza, you can't simply just make 160x more pizza out of thin air. Personally I'd just do a cubic resample if absolutely required (ideally you don't resample ofc); it's fast and straightforward. Edit: serves me right for posting, I gotta get off this site.
- superjan 9mo ago
  Maybe the following helps: if you have a an analog signal where there are no frequencies above 22.05 khz, it is in principle possible to sample it at 44.1 khz and then perfectly reconstruct the original signal from those samples. You could also represent the same analog signal using 48 khz samples. The key to resampling is not finding a nice looking interpolation, but rather one that corresponds to the original analog signal.
- deleted 9mo ago
  [deleted]
HelloUsername 9mo ago
(2022)
amlib 9mo ago
I wonder if this problem could be "solved" by having some kind of "dual mode" DACs that can accept two streams of audio at different sample rates, likely 44.1khz and 48khz, which are converted to analog in parallel and then mixed back together at the analog output. Then at the operating system level rather than mixing everything to a single audio stream at a single sample rate you group each stream that is at or a multiple of either 44.1khz or 48khz and then finally sends both streams to this "dual dac", thus eliminating the need to resample any 44.1khz or 48khz stream, or even vastly simplifying the resample of any sample rate that is a multiple of this.
- PunchyHamster 9mo ago
  > I wonder if this problem could be "solved" by having some kind of "dual mode" DACs that can accept two streams of audio at different sample rates, likely 44.1khz and 48khz, which are converted to analog in parallel and then mixed back together at the analog output. You'd just resample both at 192kHz and run it into 192kHz DAC. The "headroom" means you don't need to use the very CPU intensive "perfect" resample.
adzm 9mo ago
As a real world example, on Windows, unless you take exclusive access of the audio output device, everything is already resampled to 48khz in the mixer. Well, technically it gets resampled to the default configured device sample rate, but I haven't seen anything other than 48khz in at least a decade if ever. Practically this is a non-issue, though I could understand wanting bit-perfect reproduction of a 44.1 khz source.
klaff 9mo ago
I'm kinda shocked that there's no discussion of sinc interpolation and adapting it's theoretical need for infinite signals to some finite kernel length. For a sampled signal, if you know the sampling satisfied Nyquist (i.e., there was no frequency content above fs/2) then the original signal can be reproduced exactly at any point in time using sinc interpolation. Unfortunately that theoretically requires an infinite length sample, but the kernel can be bounded based on accuracy requirements or other limiting factors (such as the noise which was mentioned). Other interpolation techniques should be viewed as approximations to sinc. Sinc interpolation is available on most oscilloscopes and is useful when the sample rate is sufficient but not greatly higher than the signal of interest.
- oakwhiz 9mo ago
  Sinc interpolation is somewhat resource-intensive / latency inducing which is why it's not used as frequently in realtime audio.
  - klaff 9mo ago
    Agree, but it's like knowing a limit imposed by the 2nd law of thermodynamics. You can't do better and if you get close enough you are done.
observationist 9mo ago
If you're taking something from 44.1 to 48, only 91.875% of the data is real, so 8.125% of the resulting upsampled data is invented. Some of it will correlate with the original, real sound. If you use upsampling functions tuned to features of the audio - style, whether it's music, voice, bird recordings, NYC traffic, known auditorium, etc, you can probably bring the accuracy up by several percent. If the original data already has the optimizations, it'll be closer to 92%. If it's really good AI upsampling, you might get qualitatively "better" sounding audio than the original but still technically deviates from the original baseline by ~8%. Conversely, there'll be technically "correct" upsampling results with higher overall alignment with the original that can sound awful. There's still a lot to audio processing that's more art than science.
fulafel 9mo ago
> In reality, the amount of precision that can actually be "heard" by the human ear probably lies between 18 and 21 bits; we don't actually know, because it's impossible to test. This sounds contradictory - what would be the precision that can be heard in a test then?
- yeasku 9mo ago
  [dead]
functionmouse 9mo ago
This article cleared up so many long-standing questions for me. THANK YOU for sharing!!!
yeasku 9mo ago
[dead]
somat 9mo ago
> once we have reduced errors to below the noise floor they are inaudible by definition. Makes me think of GPS where the signal is below the noise floor. Which still blows my mind, real RF black magic.
nivea3066 9mo ago
For those looking to delve into this topic more, the term of art is ASRC: Asynchronous Sample Rate Conversion.

51 points by brewmarche 9mo ago

> it's probably worth avoiding the resampling of 44.1 to 48 kHz Ehhm, yeah, duh? You don't resample unless there is a clear need, and even then you don't upsample and only downsample, and you tell anyone that tries to convince you otherwise to go away and find the original (analog) source, so you can do a proper transfer.

mort96 9mo ago
If only it was that simple T_T I'm working on a game. My game stores audio files as 44.1kHz .ogg files. If my game is the only thing playing audio, then great, the system sound mixer can configure the DAC to work in 44.1kHz mode. But if other software is trying to play 48kHz sound files at the same time? Either my game has to resample from 44.1kHz to 48kHz before sending it to the system, or the system sound mixer needs to resample it to 48kHz, or the system sound mixer needs to resample the other software from 48kHz to 44.1kHz. Unless I'm missing something?
- mort96 9mo ago
  And actually, why do we have both 48kHz and 44.1kHz anyway? If all "consumer grade high quality audio" was in 44.1kHz (or 48kHz) we probably could've avoided resampling in almost all circumstances other than professional audio contexts (or for already low quality audio like 8kHz files). What benefit do we get out of having both 44.1 and 48 that outweighs all the resampling it causes?
  - deleted 9mo ago
    [deleted]
  - zamadatix 9mo ago
    As far as I understood, both rates ultimately come from trying to map to video standards of the time. 44.1 kHz mapped great to reusing analog tape of the time, 48 kHz mapped better to digital clocking and integer multiples of video standards while also having a slightly wider margin on oversampling the high frequency. 44.1 kHz never really went away because CDs continued using it, allowing them to take any existing 44.1 kHz content as well as to fit slightly more audio per disc. At the end of the day, the resampling between the two doesn't really matter and is more of a minor inconvenience than anything. There are also lots of other sampling rates which were in use for other things too.
  - CharlesW 9mo ago
    > And actually, why do we have both 48kHz and 44.1kHz anyway? Those two examples emerged independently, like rail standards or any number of other standards one can cite. That's really just the top of the rabbit-hole, since there are 8-20 "standard" audio sample rates, depending how how you count. This isn't really a drawback, and it does provide flexibility when making tradeoffs for low bitrates (e.g. 8 kHz narrowband voice is fine for most use cases) and for other authoring/editing vs. distribution choices.
    - Joeboy 9mo ago
      > This isn't really a drawback But, that's only true because people freely resample between them all the time and nobody knows or cares about it.
    - pixelpoet 9mo ago
      The nice thing about standards is, there are so many from which to choose! :)
  - sneak 9mo ago
    CDs used 44.1, DAT and DVDs used 48. That’s it. Most stuff on the internet ripped from CD is 44.1. 48 is getting more common. We’re like smack in the middle of the 75 year transition period to 48kHz. For new projects, I use 48, because my mics are 32bit (float!)/48kHz.
  - brudgers 9mo ago
    Perhaps the benefit we get is access to existing recordings? 44.1khz exists because it was the lowest technically practical speed and was an optimization for processing speed and storage space. 48khz exists because it syncs with video easily — I’ve also heard it allows for more tolerance in the anti-aliasing filter.
    - cormorant 9mo ago
      > 48khz exists because it syncs with video easily I guess meaning 24fps video? Because 44100 is already a multiple of 25, 30, 50, and 60.
      - adgjlsfhk1 9mo ago
        48k also supports 24, 48, 120, and 240 which are all nice to haves.
      - brudgers 9mo ago
        24fps is where the money is because 24fps is the standard for film.
  - PunchyHamster 9mo ago
    technically we could use 40kHz and just upsample, the extra frequency over 40kHz is basically leeway to make analog part possible/cheap, but it is not technically needed in the signal the first CD player didn't had compute power to upsample perfectly but modern devices certainly do.
    - mort96 9mo ago
      AFAIU, 40kHz exactly wouldn't really work, if your goal is to represent 0Hz-20kHz: in order to avoid aliasing, you need a low pass filter to remove all frequency content above half your sample rate, and no filter is infinitely hard (and you generally want to give the filter a decent range of frequencies to work with). If you want to start your low pass filter at 20kHz, you want it to end (i.e reach practically -∞dB) at a few kHz above 20kHz. If you used a sample rate of exactly 40kHz, you would need your low pass filter to reach -∞dB at 20kHz, meaning it'd have to start somewhere in the audible region. Though this is just my understanding. Maybe I'm wrong.
  - rerdavies 9mo ago
    > Why do we have both 48kHz and 44.1kHz anyway Because of greed. Early audio manufacturers (SONY notably) used 48kHz for profession-grade audio equipment, that would be used in studios or TV stations, and degraded 44.1khz audio for consumer devices. Typically you would pay an order of magnitude more for the 48kHz version of the hardware. 48khz is better for creating and mixing audio. You cannot practically mix audio at 44.1khz without doing very slight damage to audible high frequencies. But enough to make a difference. If you were creating for consumer devices, you would mix at 48Khz, and then downsample to 44.1khz during final mastering, since conversion from 48kHz to 44.1kHz can be done theoretically (and practically) perfectly. (Opinions of the OP notwithstanding). I think it's safe to say that the 44.1kHz sampling rate was maliciously selected specifically because it is just low enough that perfect playback is still possible, but perfect mixing is practically not possible. And obviously maliciously chosen to be a rate with no convenient greatest common denominator with 48Khz, which would have allowed easy and cheap perfect realtime resampling. Had Sony chose 44.0kHz, it would be trivially easy to do sample rate conversion to 48Khz in realtime even with primitive hardware available in the late 1970s. That extra .1kHz is transparently obvious malice and greed in plain sight. Presumably SONY would sell you the software or hardware to perform perfect non-realtime conversion of audio from 48khz to 44.1khz for a few tens of thousands of dollars. Not remotely subtle how greedy all of this was. There has been no serious reason to use 44.1kHz instead of 48kHz for about 50 years, at least from a technology point of view. (And no real reason to EVER use 44.1khz instead of 48kHz other than GREED).
    - brendyn 9mo ago
      Are you able to share evidence for this?
      - rerdavies 9mo ago
        What would you consider evidence? Emails between standards committee members agreeing to collude in order to screw pro-audio customers? The evidence is: why on earth would anyone on a standards committee choose 44.1kHz, instead of 44.0kHz? The answer: 44.1kHz was transparently obviously chosen to make it impossible to perform on-the-fly rate conversions. The mathematics of polyphase rate converters was perfectly well understood at the time these standards were created.
- zamadatix 9mo ago
  That's a clear need IMO, but it'd be slightly better if the game could have 48 kHz audo files and downsampled them to 44.1 kHz playback than the other way around (better to downsample than upsample).
  - adgjlsfhk1 9mo ago
    They're both fine (as long as the source is band limited to 20khz which it should be anyway).
    - adrian_b 9mo ago
      The analog source is never perfectly limited to 20 kHz because very steep filters are expensive and they may also degrade the signal in other ways, because their transient response is not completely constrained by their amplitude-frequency characteristic. This is especially true for older recordings, because for most newer recordings the analog filters are much less steep, but this is compensated by using a much higher sampling frequency than needed for the audio bandwidth, followed by digital filters, where it is much easier to obtain a steep characteristic without distorting the signal. Therefore, normally it is much safer to upsample a 44.1 kHz signal to 48 kHz, than to downsample 48 kHz to 44.1 kHz, because in the latter case the source signal may have components above 22 kHz that have not been filtered enough before sampling (because the higher sampling frequency had allowed the use of cheaper filters) and which will become aliased to audible frequencies after downsampling. Fortunately, you almost always want to upsample 44.1 kHz to 48 kHz, not the reverse, and this should always be safe, even when you do not know how the original analog signal had been processed.
      - PunchyHamster 9mo ago
        yeah but you can record it in 96kHz, then resample it perfectly to 44.1 (hell, even just 40) in digital domain, then resample it back to 48kHz before sending to DAC
        adrian_b 9mo ago
- zipy124 9mo ago
  you're not missing something. You can re-sample them safely as stated by the author. They simply state you should check the re-sampler as: > Although this conversion can be done in such a way as to produce no audible errors, it's hard to be sure it actually is. That is, you should verify the re-sampler you are using or implement yourself in order to be sure it is done correctly, and that with todays hardware it is easily possible.
- Veliladon 9mo ago
  Getting pristine resampling is insanely expensive and not worth it. If you have a mixer at 48KHz you'll get minor quantization noise but if it's compressed already it's not going to do any more damage than compression already has.
- ZeroConcerns 9mo ago
  DAC frequency and the audio format requirements for whatever you supply to your platform audio API have literally nothing to do with each other. My reply was from an audio mastering perspective.
- Joeboy 9mo ago
  > Unless I'm missing something? I suppose the option you're missing is you could try to get pristine captures of your samples at every possible sample rate you need / want to support on the host system.
- zelphirkalt 9mo ago
  Is this not the job of the operating system or its supporting parts, to deal with audio from various sources? It should not be necessary to inspect the state of the OS your game is running on, to know what kind of audio you can playback. In fact, that could even be considered spying on things you shouldn't. Maybe the OS or its sound system does not abstract that from you and I am wrong about the state of OS in reality, but this seems to me like a pretty big oversight, if true. If I extrapolate from your use-case, then that would mean any application performing any playback of sound, needs to inspect whether something else is running on the system. That seems like a pretty big overreach. As an example, lets say I change frequency in Audacity and press the play button. Does Audacity now go and inspect, whether anything else on my system is making any sound?
  - Joeboy 9mo ago
    > Is this not the job of the operating system or its supporting parts, to deal with audio from various sources I think that's the point? In practice the OS (or its supporting parts) resample audio all the time. It's "under the hood" but the only way to actually avoid it would be to limit all audio files and playback systems to a single rate.
    - zelphirkalt 9mo ago
      I don't understand then, why they need to deal with that when making a game, unless they are not satisfied with the way that the OS resamples under the hood.
      - Joeboy 9mo ago
        My reading is not that they're saying it's something they necessarily have deal with themselves, but that it's something they can't practically avoid.
        rerdavies 9mo ago
        But they CAN practically avoid it. lol. Just let the system do it for them.
- adzm 9mo ago
  You are right; the system sound mixer should handle all resampling unless you explicitly take exclusive control of the audio device. On Windows at least, this means everything generally gets resampled to 48khz. If you are trying to get the lowest latency possible, this can be an obstacle... on the order of single digit milliseconds.
bob1029 9mo ago
If 44.1kHz is otherwise sufficient but you have a downstream workflow that is incompatible, there are arguments for doing this. It can be done with no loss in quality. From an information theory perspective, this is like putting a smaller pipe right through the middle of a bigger one. The channel capacity is the only variable that is changing and we are increasing it.
- anonymars 9mo ago
  This isn't the whole picture because they aren't multiples of each other, so there will be interpolation/jitter For example if you watch a 24fps film on a 60fps screen, in contrast to a 120fps screen
  - mort96 9mo ago
    That's not how audio works. PCM data at some sample rate with infinite precision samples perfectly encodes all the audio data of all frequencies up to half the sample rate. Resampling is a theoretically lossless operation when the frequency content of the audio fits within half the sample rate of both the source and the destination sample rates (which will always be true when resampling to a higher sample rate, FWIW). The issues are that 1) resampling has a performance and latency cost, 2) better resampling has a higher performance and latency cost
zipy124 9mo ago
That seems a rather shallow - and probably incorrect - reading of the source. This is an efficiency and trust trade off as noted: > given sufficient computing resources, we can resample 44.1 kHz to 48 kHz perfectly. No loss, no inaccuracies. and then further > Your smartphone probably can resample 44.1 kHz to 48 kHz in such a way that the errors are undetectable even in theory, because they are smaller than the noise floor. Proper audio equipment can certainly do so. That is you don't need the original source to do a proper transfer. The author is simply noting > Although this conversion can be done in such a way as to produce no audible errors, it's hard to be sure it actually is. That is that re-sampling is not a bad idea in this case because it's going to have any sort of error if done properly, it's just that the Author notes you cannot trust any random given re-sampler to do so. Therefore if you do need to resample, you can do so without the analog source, as long as you have a re-sampler you can trust, or do it yourself.
- sgerenser 9mo ago
  Speaking of a resampler you trust, I’ve had good experience with libsamplerate (http://www.mega-nerd.com/SRC/ http://www.mega-nerd.com/SRC/), which as of 2016 is BSD licensed.
brudgers 9mo ago
A very common clear need is incorporating 44.1khz audio sourcesinto video. 48khz is 48khz because 48khz divided by 24fps, 25fps, or 30fps is an integer (and 44.1khz is not). Also, for decades upsampling on ingest and downsampling on egress has been standard practice for DSP because it reduces audible artifacts from truncation and other rounding techniques. Finally, most recorded sound does not have an original analog source because of the access digital recording has created…youtube for example.

AshamedCaptain 9mo ago

Lots of Live/Audigy era Creative sound cards would resample everything to 48kHz, with probably one of the worst quality resamplers available, to the chagrin of all bitperfect fanatics... still probably one of their best selling sound cards. I.e. no one cares.

Tanoc 9mo ago
I had a Soundblaster Live! Gold card back in the day, and I would route my record player or stereo through it so I could use a visualizer on my computer. You could hear the digital noise that was introduced on the highhats. And the source for the sound was a late '70s era Realistic system where everything was analogue. I never knew it was because of the soundcard. I'd always just chalked it up to either Windows XP or VLC doing something.
- AshamedCaptain 9mo ago
  Unlikely to be this issue since this is about resampling of (e.g. from a CD's native) 44.1kHz PCM to 48kHz, and has nothing at all to do when recording since you'd most likely record at 48kHz and play at 48kHz (no HW resampling involved).

everfrustrated 9mo ago

Changing the sample rate of audio only affects the frequency range. All audio signal is _perfectly_ represented in a digital form. I am ashamed to admit this took me a long time to properly understand. For further reading I'd recommend: https://people.xiph.org/~xiphmont/demo/neil-young.html https://people.xiph.org/~xiphmont/demo/neil-young.html https://www.youtube.com/watch?v=cIQ9IXSUzuM https://www.youtube.com/watch?v=cIQ9IXSUzuM

pxndxx 9mo ago
Sampling does not lose information below the Nyquist limit, but quantization does introduce errors that can't be fixed. And resampling at a different rate might introduce extra errors, like when you recompress a JPEG.
- brudgers 9mo ago
  In the audio world, quantization is usually discussed in terms of bit-depth rather than sample rate.
  - mort96 9mo ago
    Yeah, they know and their comment reflects that knowledge. They're saying that if we had infinite bit depth, we could arbitrarily resample anything to anything as long as the sample rate is above the Nyquist frequency; however we don't have an infinite bit depth, we have a finite bit depth (i.e the samples are quantized), which limits the dynamic range (i.e introduces noise). This noise can compound when resampling.
    - adgjlsfhk1 9mo ago
      The key point is that even with finite bit depth (as long as you dither properly), the effect of finite bit depth is easily controlled noise of program chosen spectrum. i.e. as long as your sampling isn't doing anything really dumb, the noise introduced by sampling is well below noise floor.

cozzyd 9mo ago

Naively I would upsample by 4-8 by 0 stuffing and low pass filtering and then interpolating. That can't be that bad, can it?

MontagFTB 9mo ago

> We do [cubic curve fitting] all the time in image processing, and it works very well. It would probably work well for audio as well, although it's not used -- not in the same form, anyway -- in these applications. Is there a reason the solution that "works very well" for images isn't/can't be applied to audio?

amlib 9mo ago
AFAIK it introduces harmonic distortion
- pixelpoet 9mo ago
  I'd love to know more about this, do you perhaps have any refs? Thanks
  - amlib 9mo ago
    Not an expert in this field, just a scrub, so I can't really give you much. There is this website that has painstakingly compares many resampling algorithms from all sorts of software: https://src.infinitewave.ca https://src.infinitewave.ca Try it's mirror if you can't access it: https://megapro17.github.io/src/index.html https://megapro17.github.io/src/index.html The only one that says it is a cubic interpolation is the "Renoise 2.8.0 (cubic)" one, the spectrogram isn't very promising with all sorts of noise, intermodulation and aliasing issues. And, by switching to the 1khz tone spectrum view you can see some harmonics creeping up. When I used to mess with trackers I would sometimes chose different interpolations and bicubic definitely still colored the sound, with sometimes enjoyable results. Obviously you don't want that as a general resampler...
    - butterknife 9mo ago
      Just to note that this site hasn't been updated for a while. Much better, more modern and with automated upload analysis site would be [1] although it is designed for finding the highest fidelity resampler rather than AB comparisons. [1] https://src.hydrogenaudio.org https://src.hydrogenaudio.org

bobbylarrybobby 9mo ago

Where does dither fit into this picture?

mistrial9 9mo ago
dither is not compression or re-sampling. Dither is a means to change data so that it appears similar to original data to a human AFAIK source- wrote dithering code for digital images
- pixelpoet 9mo ago
  Dithering is about trading spatial / time resolution for reduced average quantisation error.
  - mistrial9 9mo ago
    The word dither -- evidently, the term dithering originally referred to a trembling or erratic movement, from the Middle English didderen, meaning "to tremble." further, the halftone technique developed in the 1880s by Georg Meisenbach — breaking images into dots to simulate shades of gray — was called autotype, not dithering. The term dithering was later adopted in digital imaging and computing, particularly in the 1960s, when engineers applied the concept of adding noise to reduce color banding.
Sesse__ 9mo ago
You generally want to dither when (before) you quantize, unless you have so much headroom that it doesn't matter. E.g., if you're converting 44.1/16 to 48/16 (which involves quantizing each sample from an intermediate higher-precision result), you probably want to dither, but if you're converting 44.1/24 to 48/24, you probably won't need to care since you don't really care about whether your effective result is 24 or 24.5–25 bits.
ErroneousBosh 9mo ago
That's more of a bit depth rather than bit rate thing. I was surprised to find that going from 16 to 8 bits by simply truncating gave really obvious artifacts on certain sounds (a sampled 808 kick for example had a distinct BZZZEEEOOOOWWWW sound, quite prominent), and even really simple triangular noise dithering made it go away. It did mean the playback was more noisy but it was less obvious. As an aside, G.711 codecs use a kind of log scale with only four bits of signal but small signal values use much smaller bits.

pixelpoet 9mo ago

I'm not sure I understand the "just generate it" perspective. If you want to generate a much higher sampling rate signal that has a common multiple of your input and output sampling rate, "just generating it" is going to involve some kind of interpolation, no? Because you're trying to make data that isn't there. If you want to change the number of slices of pizza, you can't simply just make 160x more pizza out of thin air. Personally I'd just do a cubic resample if absolutely required (ideally you don't resample ofc); it's fast and straightforward. Edit: serves me right for posting, I gotta get off this site.

superjan 9mo ago
Maybe the following helps: if you have a an analog signal where there are no frequencies above 22.05 khz, it is in principle possible to sample it at 44.1 khz and then perfectly reconstruct the original signal from those samples. You could also represent the same analog signal using 48 khz samples. The key to resampling is not finding a nice looking interpolation, but rather one that corresponds to the original analog signal.
deleted 9mo ago
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HelloUsername 9mo ago

(2022)

amlib 9mo ago

I wonder if this problem could be "solved" by having some kind of "dual mode" DACs that can accept two streams of audio at different sample rates, likely 44.1khz and 48khz, which are converted to analog in parallel and then mixed back together at the analog output. Then at the operating system level rather than mixing everything to a single audio stream at a single sample rate you group each stream that is at or a multiple of either 44.1khz or 48khz and then finally sends both streams to this "dual dac", thus eliminating the need to resample any 44.1khz or 48khz stream, or even vastly simplifying the resample of any sample rate that is a multiple of this.

PunchyHamster 9mo ago
> I wonder if this problem could be "solved" by having some kind of "dual mode" DACs that can accept two streams of audio at different sample rates, likely 44.1khz and 48khz, which are converted to analog in parallel and then mixed back together at the analog output. You'd just resample both at 192kHz and run it into 192kHz DAC. The "headroom" means you don't need to use the very CPU intensive "perfect" resample.

adzm 9mo ago

As a real world example, on Windows, unless you take exclusive access of the audio output device, everything is already resampled to 48khz in the mixer. Well, technically it gets resampled to the default configured device sample rate, but I haven't seen anything other than 48khz in at least a decade if ever. Practically this is a non-issue, though I could understand wanting bit-perfect reproduction of a 44.1 khz source.

klaff 9mo ago

I'm kinda shocked that there's no discussion of sinc interpolation and adapting it's theoretical need for infinite signals to some finite kernel length. For a sampled signal, if you know the sampling satisfied Nyquist (i.e., there was no frequency content above fs/2) then the original signal can be reproduced exactly at any point in time using sinc interpolation. Unfortunately that theoretically requires an infinite length sample, but the kernel can be bounded based on accuracy requirements or other limiting factors (such as the noise which was mentioned). Other interpolation techniques should be viewed as approximations to sinc. Sinc interpolation is available on most oscilloscopes and is useful when the sample rate is sufficient but not greatly higher than the signal of interest.

oakwhiz 9mo ago
Sinc interpolation is somewhat resource-intensive / latency inducing which is why it's not used as frequently in realtime audio.
- klaff 9mo ago
  Agree, but it's like knowing a limit imposed by the 2nd law of thermodynamics. You can't do better and if you get close enough you are done.

observationist 9mo ago

If you're taking something from 44.1 to 48, only 91.875% of the data is real, so 8.125% of the resulting upsampled data is invented. Some of it will correlate with the original, real sound. If you use upsampling functions tuned to features of the audio - style, whether it's music, voice, bird recordings, NYC traffic, known auditorium, etc, you can probably bring the accuracy up by several percent. If the original data already has the optimizations, it'll be closer to 92%. If it's really good AI upsampling, you might get qualitatively "better" sounding audio than the original but still technically deviates from the original baseline by ~8%. Conversely, there'll be technically "correct" upsampling results with higher overall alignment with the original that can sound awful. There's still a lot to audio processing that's more art than science.

fulafel 9mo ago

> In reality, the amount of precision that can actually be "heard" by the human ear probably lies between 18 and 21 bits; we don't actually know, because it's impossible to test. This sounds contradictory - what would be the precision that can be heard in a test then?

yeasku 9mo ago
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functionmouse 9mo ago

This article cleared up so many long-standing questions for me. THANK YOU for sharing!!!

yeasku 9mo ago

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somat 9mo ago

> once we have reduced errors to below the noise floor they are inaudible by definition. Makes me think of GPS where the signal is below the noise floor. Which still blows my mind, real RF black magic.

nivea3066 9mo ago

For those looking to delve into this topic more, the term of art is ASRC: Asynchronous Sample Rate Conversion.