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24/192 music downloads are silly

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Re: 24/192 music downloads are silly

#161

24bit absolutely makes sense in the age of streaming devices pushing sound to digital inputs on powered speakers. Simply for convenience reasons I want to adjust the playback volume already at my phone. I have no analog volume control (or rather, it's a configuration knob you set up once) If you play back a sample at 1/8 the volume, you still have 21bits of range from a 24bit sample. Playing a 16bit sample at 1/8 vol…

A 24bit digital format doesn't mean the DAC actually has 24bit of SNR. Hell, most DACs in computers, phones and the like struggle with producing 16 bits. Not to mention that even if you had that headroom, when approaching the threshold of hearing would mean that you wouldn't be able to tell a difference at lower volumes.

Re: 24/192 music downloads are silly

#162
post #41

I agree with the silliness of 192kHz, but not 24-bits. Here is why: In typical PCM recordings, like CDs, mid-range frequencies (e.g. 1kHz to 4kHz) are recorded with lower amplitudes because our ears are more sensitive to them. Sampling theory is correct and 16-bits can reproduce any waveform with ~100dB of range, however, in a complex waveform consisting of low, mid and high frequencies, the mid- and hi-range frequen…

192kHz may not be as silly as you think: http://www.ncbi.nlm.nih.gov/pubmed/10848570

Thanks for posting that! However, it's worth point out that that study has its detractors: http://en.wikipedia.org/wiki/Hypersonic_effect

I find the intermodulation argument a convincing one - it's hard to not have it affect the test, and if they haven't taken specific steps to avoid it then it would be easy to join a large amount of tests that have fallen prey to it.

I note, however, that Wikipedia doesn't mention any specifically brain-scan-based studies that counter it. If you know of any more I'd be very interested in hearing about it.

Re: 24/192 music downloads are silly

#163
I think most effective way to improve sound quality is to get a good DAC as close as possible to the output. Headphone amps with integrated DACs do wonders for little money. When possible go for XLR on the last mile to the speakers (good neutral studio speakers) to cancel out distortion from external electromagnetic pulses. To me differential, or balanced signalling is still the most clever, yet so simple, analog information preserving method I ever heard of. http://en.wikipedia.org/wiki/Differential_signaling

Also; of cause 24bit increases the resolution of the signals amplitude, that's exactly what it does by definition, that it sets the noise floor is only a function of that. To stress the visual analogy: it's like looking at 16bit images with banding and then on 24 bit images. Of cause the effect is not very pronounced but it's there. Especially if you have a high dynamic recording of a soundscape but you would like to zoom in into a certain volume range like that of the human voice it's good to have that extra resolution not only in a studio environment. Think about it like developing a picture from RAW into JPEG to stress this analogy again, this doesn't make a difference for compressed pop music but for uncompressed recordings of live performances with analog instruments or natural soundscapes it does. You can then choose at which volume range your most at home for listening and do the compression or not.

I like to think about 24bit audio like having access to the RAW files of images, it doesn't matter in most cases, and most likely you should leave the mixing to professionals artists for the intended effect, but it also enables you to experience the sound in many more different ways.

Re: 24/192 music downloads are silly

#164
post #150

Earlier quoted context omitted.

> being open to accepting a mistaken belief. This is odd. Do you think "mistaken beliefs" are clearly labeled so you can reject them without examination or experimentation?

A belief that relies on known falsehoods is plainly mistaken.

I understand your meaning but that's not what I'm asking.

Re: 24/192 music downloads are silly

#165

Earlier quoted context omitted.

Wasn't a cello (I keep remembering hearing "cello attack") supposed to be problematic on mp3?

MP3 has specific algorithmic weaknesses above 16khz[0] which even the highest legal bitrate can't cover up… sometimes. It's actually easiest to hear it with cymbals, or the old LAME sample "fatboy". You can just not use MP3 though. It's 2014! Use AAC! [0] http://wiki.hydrogenaud.io/index.php?title=LAME_Y_switch

I don't understand why everyone doesn't use Opus for anything. Damn thing is free, beats AAC for psychoacoustics, beats Speex for latency...

Re: 24/192 music downloads are silly

#166

Fascinating article. Is there any reason to believe that upsampling an audio file should produce a better analog signal than playing it at its native sample rate? I find that playing high quality mp3's through an upsampling DSP filter and a 24/192 DAC seems to produce a better listening experience. (As the article points out, this could be due to confirmation bias, or the filter making the music a tiny bit louder.) I…

> Intuitively it makes sense to me that the DAC sending signals twice as frequently to the headphones would produce a smoother analog signal, but is that actually true? No. Watch the Digital Media Primer and Digital Show & Tell, Monty (the article's author) explains how digital sound encoding works and why the analog signal will be reproduced beyond your ability to notice imperfections either way: http://xiph.org/vid…

The engineering reason to upsample is to simplify the analog anti-aliasing filter on the other side of the conversion by giving it a wider transition band to work with. It also means one analog filter can handle a wide range of samplerates.

masklinn is right though—from a consumer perspective, there's no real reason to care how the conversion's done. Hopefully your DAC was designed by a team that knows what they're doing and will handle whatever rate you feed it.

Of course, if you have one of those obnoxious devices that can't actually clock below 48kHz or a mixing daemon configured to run at 48kHz, you're stuck oversampling anyway, and depending on your environment that might be done anywhere between amazingly well and linear interpolation. Many years ago I was stuck playing that game on Linux—in that case, there's an audible benefit from using a better designed upsampler in your player.

But if you're equipment's not broken, it's a waste of CPU cycles.

Re: 24/192 music downloads are silly

#167
post #57
post #52

Earlier quoted context omitted.

If there's psychological phenomena that make them enjoy it more, what's the issue?

And there are jobs created for engineers who design that equipment - simultaneously recycling more money in the economy and contributing to the statistics of STEM job prospects! Everybody wins!

And I get cheap hight definition AD and DA conversors to use in my projects!

I'm certainly not complaining.

Re: 24/192 music downloads are silly

#168

I once spent ~2 hours explaining all of this (well maybe not all of it) to a friend of mine who was studying sound design at the time. He didn't believe me. I even transcoded some 24/192 FLAC Pink Floyd I had lying around and made him do a double blind test to show him that he'd prefer the slightly louder song every time, even if the louder song was 192kbps vs the FLAC. He did. He still doesn't believe me. He still t…

Maybe it depends on the song as well? Classical music is supposed to have a much greater frequency spectrum - meaning, the effects of MP3 encoding becomes apparent when played on high fidelity equipment.

It has greater dynamic range (think volume, this is the domain of the bit rate properly called bit depth) than most compressed studio-produced music but the frequency range or spectrum (think pitch, which is limited on the high end by the sampling rate/2 via the Nyquist-Shannon theorem) is not different if not also clipped off by mixing.

Re: 24/192 music downloads are silly

#169

Earlier quoted context omitted.

The resolution of both sinusoids in the example you suggest is exactly 16bits Edit: While it would be possible to represent each of those individual sinusoids in 6 and 10 bits individually, the signal you describe has the high frequency signal "riding on " the low frequency signal. you need 16 bits to represent the amplitude of that signal at e.g. t = pi/4.

2^10 + 2^6 = 1088. You only need 11 bits to not clip with that signal.

you'd need 11 bits to encode the signal at 0dBFS. Theres amplitude information stored in those other 5 zeros...

Re: 24/192 music downloads are silly

#170
post #155

Monty's example files made me realize that my computer's sound chip is probably kind of lo-fi. No, I couldn't hear anything in my headphones when I played the 30Khz/33Khz tones. In comparison, I heard something after it stopped . Somehow it was quieter during the playback. Does it make any sense that my computer is better at making no audible sound when it's asked to play a loud inaudible sound, than when it's being…

It is likely a power saving measure. There is a slightly audible click when mine turns on and off automatically, and a bit of noise when it's off.
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