[ZeroConcerns]

> 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.

[zipy124]

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]

Speaking of a resampler you trust, I’ve had good experience with libsamplerate (http://www.mega-nerd.com/SRC/), which as of 2016 is BSD licensed.

[mort96]

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?

[adzm]

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.

[mort96]

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?

[zelphirkalt]

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?

[Veliladon]

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.

[zamadatix]

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).

[Joeboy]

> 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.

[ZeroConcerns]

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.

[zipy124]

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.

[bob1029]

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]

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

[brudgers]

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.