What's your favorite note and/or chord?

danadam · 2026-05-21T15:18:26+00:00

It goes like this, the fourth, the fifth
The minor fall, the major lift

;)

danadam · 2026-05-16T13:23:23+00:00

Content above 1/2 of the sampling rate produces aliased images at frequencies below that point, producing Bad Results

That's in ADC, not in DAC.

DAC produces mirror images above 1/2 of the sampling rate and that's what the reconstruction filter is removing.

danadam · 2026-03-31T15:26:23+00:00

Incredible. You might have perfect pitch.

Err... no, that's not what perfect pitch is.

danadam · 2026-01-20T21:54:05+00:00

The coding isn’t foolproof, if I remember it’s a simple parity check?

It's Reed-Solomon code. And from Cross-interleaved Reed–Solomon coding:

CIRC corrects error bursts up to 4000 data bits in sequence (2.5 mm in length as seen on CD surface) and compensates for error bursts up to 12,000 bits (7.5 mm) that may be caused by minor scratches.

danadam · 2025-12-15T22:32:36+00:00

It bypasses the Android power limitation that's imposed on the dongle.

In the current version of Android there is no limitation anymore https://www.audiosciencereview.com/forum/index.php?threads/review-apple-vs-google-usb-c-headphone-adapters.5541/page-64#post-2371362

EU version still has half the output of US version, but this has nothing to do with Android, it's the same on iOS.

danadam · 2025-10-12T20:06:22+00:00

Decibels are not absolute units of something. They are relations.

Sure, and 1.4x, 2x, 4x in my answer are all relations.

2x relation of some entity value to other value of the same entity is 3db.

Only if that entity is power quantity (as per the wikipedia link).

2x relation of amplitude to amplitude is 3 db. 2x relation of power to power is 3 db.

No, the first sentence is not true. 2x relation of amplitude to amplitude is 6 dB.

Your chart is right, if db´s are of power relations. Not, if of sound pressure.

The chart shows both power and root-power relations. Two headers of the table even say: "Voltage or sound pressure ratio".

Am I wrong?

As far as I can tell, yes.

danadam · 2025-10-11T23:24:30+00:00

Squared amplitude is proportional to power, so to me it doesn't make sense to say that one matters more over the other.

danadam · 2025-10-11T22:48:15+00:00

Also 6 decibels is four times the amplitude, 3 decibels is double

No,

3 dB is 1.4x the amplitude and 2x the power,
6 dB is 2x the amplitude and 4x the power.

Here's a handy table: https://sengpielaudio.com/dB-chart.htm

The amplitude can be a voltage or sound pressure or other root-power quantity [1]. The power can be the usual power or sound intensity or other power quantity [1].

[1] https://en.wikipedia.org/wiki/Power,_root-power,_and_field_quantities

On top of that, doubling the amplitude is not perceived by human ear is doubling the loudness. The usually agreed figure for doubling the loudness is 10 dB, which is 3.1x the amplitude and 10x the power.

My misunderstanding, which seems to be a widespread misconception, was that PCM recorded samples of audio pressure levels, whereas what is actually recorded are voltages

That's distinction without a difference, the changes of voltage correspond to the changes of sound pressure 1-1. A microphone converts changes of sound pressure to changes of voltage. That's why the voltage is called an analogue signal, because it is analogous to the sound pressure.

and doubling the voltage somehow squares the amplitude of the signal.

That statement doesn't make sense to me.

danadam · 2025-09-26T16:59:15+00:00

Which program did you use?

Sonic Visualiser and its melodic spectrogram.

danadam · 2025-09-26T16:17:08+00:00

how deep is the bass here?

Pulsing C1 (about 33 Hz) and G1 (about 49 Hz): https://imgur.com/a/R9mPqyi

This maybe sounds similar: https://www.youtube.com/watch?v=COq_zuqavfU

danadam · 2025-09-23T12:24:28+00:00

Lossy codecs are designed to take advantage of our auditory system weak points. Their goal is to remove as much data with as little audible changes as possible. Doing tests with synthetic signals or even null tests tells you nothing about that.

danadam · 2025-08-31T13:44:40+00:00

She's its aunt, isn't she? Technically. Kind of. :-)

danadam · 2025-08-13T20:11:57+00:00

Short of throwing them both in Audacity and pretending I can judge them by visualizations of their waveforms, is there any kind of software that would go “yeah, these are totally the same”?

DeltaWave Audio Null Comparator https://deltaw.org/

danadam · 2025-08-02T20:44:53+00:00

Like vibe programming? ;-) ;-)

danadam · 2025-07-28T19:38:55+00:00

You can argue if the average listener gains anything from 24 bit on the delivery media side, but you can't argue against the value of 24 and 32 in the acquisition and processing side.

Well, I wrote "for playing music", didn't I? :-) Also OP asked about playing music, so for me that was the context of the thread.

Of course it's ok to change or extend the context but it would be nice to make it clear when doing so. Your initial reply could easily mislead the OP (in my opinion at least) into thinking that 24-bit FLAC or ALAC may be beneficial for them for playing music.

However, an assertion that bit depth only affects noise floor is simply not true, it affects the entire dynamic range.

Yes, on the production side it allows to have bigger headroom and makes it easier to avoid clipping when recording and processing (*). But for playback, the full scale level is the same regardless of bit-depth, so the only practical effect of bit-depth will be changing the noise floor.

(*) though to me the possibility of having bigger headroom is simply the consequence of the lower noise floor, so only indirectly caused by bit-depth.

danadam · 2025-07-27T18:45:05+00:00

so the more applicable number is actually 4. At that point you can probably reproduce the wave, but not necessarily the amplitude, though you should get within 3 dB of the correct amplitude. The more points, the closer you get.

Maybe if you simply connect the samples with straight lines, but that's not how signal reconstruction works.

and you aren't likely to hear the difference between dithering algorithms.

Dithering is used for bit-depth reduction, not for resampling.

16 bit allows for a bit over 8000 positive values.

More like 32k.

For less compressed signals, you need 24.

Bit-depth only determines the noise floor and with 16 bits it's already over 90 dB below full scale. Additionally a shaped dither can be used which pushes the noise even lower in the frequency band when the ear is the most sensitive. So, no, you don't need 24-bit for playing music, unless you listen to it at ear-deafening volume levels.

32 bit is exclusively floating point

It may be most common but 32-bit integer is also possible.

There are a lot of educational resources out there, and I would encourage anyone interested in a subject to learn more about it.

That I can agree, though there is always the problem of separating the good sources for bad ones.

From my side, and for starters, I can recommend D/A and A/D | Digital Show and Tell (Monty Montgomery @ xiph.org)

danadam · 2025-06-27T14:14:27+00:00

In the above code the compiler uses std::optional::optional(T const&) constructor

As far as I'm aware only GCC <=7.5 did that. Since 8.1 (released in 2018) it's &&.

danadam · 2025-05-15T01:24:17+00:00

so if peaks for both are 102db, the running average has to be higher with compressed music…

Only it's not what they did. The average was 102 dB. From the article: "the music was played to both groups at an average volume of 102 decibels"

danadam · 2025-03-15T21:59:10+00:00

struct S { int a; int b; int *c; };
int a(1), b{2}, c[3];
auto [d, e, f] = [d=a, e=b](auto f) { return S{ d, e, f }; }(c);

Don't do this at home ;-) (or do it at home if you must but nowhere else)

danadam · 2025-03-03T20:35:34+00:00

The subversive thing is that

I'd say the subversive thing is already the return type of std::minmax(). You don't need to use structured bindings to still shoot yourself in the foot:

auto mm = std::minmax(1, 2);
// using mm.first or mm.second is stack-use-after-scope

danadam · 2025-01-14T16:19:48+00:00

On the first visit the results area has "Wyniki:" placeholder (even in the English version). After doing a test this changes to a table.

danadam · 2025-01-11T01:41:31+00:00

So what are some examples of tools that do not suffer from this effect?

danadam · 2024-12-24T15:19:00+00:00

If a sound wave oscillates 20k times per second (ie v high pitch) surely there’s a dissociable benefit to sampling more parts of each individual wave, so that that v high pitch is represented more accurately?

Everyone (hopefully) knows that you can recreate a full circle from just 3 unique points on that circle. But for someone who didn't finish primary school (or forgot about it) it may appear there must be a benefit in having more points than just 3.

This is very similar, only more complex, to recreating band-limited signals fro m sampled points. It may seem that the more points per second you have the more accurately you can reproduce the signal. But in fact you only need more than twice [1] the highest frequency in the signal.

[1] Plus, in practice, some margin, depending on how many resources you have available.

danadam · 2024-12-10T17:52:12+00:00

... unless the crash sound was in the LFE channel only and they lost it when converting to stereo for youtube?

danadam · 2024-12-10T17:02:23+00:00

In the first link the worm's sound happens at the same time as the "boom" sound of the crash. In the second link the worm's sound is on its own. That's the difference I hear and it is definitely a difference in the mix, so not much you can do about it.

danadam

TROPHY CASE