Does pono's 24-bit audio matter? Hear for yourself.

When you apply negative gain to the 16-bit signal, it has less amplitude in absolute terms, but it should have all the resolution of a whole 16-bit sample in that space.

The point is that whether that 16-bit signal is represented with the lower 4 bits of a 16-bit sample, or the lower 12 bits of a 24-bit sample, you can scarcely hear anything at all, let alone a qualitative difference between the signals.

192KHz is not a higher quality format, it is a production format. The purpose is to reduce the cost and finality of anti-aliasing filters when sampling the signal, it is just cheaper to manufacture. There's a good reason for this being the standard in professional audio, they need to buy a LOT of audio interfaces, many studios will have thousands of such inputs.

Thanks to the basic laws governing signals, we know with near certainty that not only is 192KHz overkill, but so is 48KHz, and so is 44.1. Without significant new evidence showing humans hearing signals with frequencies greater than 24KHz, you will not make any convincing argument as to why we should go with any sample rate higher than 48KHz for human listening.

As for 24-bit, it is another production interchange format. It's there so that you don't need to stand around adjusting gain knobs on audio interfaces so that you get decent fidelity but also don't clip. With 24-bit you can just sample your audio once, and assuming it's within a reasonable range, you can adjust the gain in the discrete signal. There is some indication that 16-bit sampling is less than completely ideal. The very best ears in humanity(newborn ears) distinguish about 21 bits in the safe ranges of amplitude, 24-bit may make sense in an audio system for newborn babies.

I feel that in 2014, music should be released in the best format available from the studio, and if you want a crappier version so that you can shove only 100 songs on your iWhatever, that's your choice.

Wastes space? In an age where we stream gigabyte movies? Do we need to remind the author that we're no longer in 1982 and that we have evolved beyond floppy disks?

This is the main problem with audiophiles.

Improve your equipment until it makes your music awesome, then just enjoy it.

I always like Alan Parson's quote, audiophiles use your music to listen to their equipment

It isn't equivalent enough to real music, and the human auditory sensory perception and recall systems just aren't perfect enough to make accurate judgements in those test cases. When testing you end up looking for discrepancies you can identify with words and identifiable momentary observations, things like "I can definitely hear this passage the notes are more muddled together in the compressed file." But you miss so many things that show up as minor feelings of unidentifiable hunches. Of course, you could change the format of the test, but even still, "identifying" is not "listening."

Things you can not identify, or talk about, or remember, or form sentences about, can still impact your musical experience. Our ears and brains are complex, and the range of input that is perceived subconsciously is astounding. We accept this for vision, for taste, for emotions, for memory, but for some reason not for audio. The simple feelings that you can't put your finger on can be important to the experience even if they can't be identified. But if you start to talk about "realism" and "emotion" and "feeling," you are immediately blacklisted by the empirical measurement mafia.

There is so much truth to the fact that scientific measurements and empiricism are important in determining the decisions you make about your audio storage and listening. You shouldn't pay money for things you can prove won't make a difference, and there are so many things out there that fit that bill. But we shouldn't throw out entire possibilities of discussion just because they influence parts of our experience that are subconscious or unidentifiable in an A/B test.

Again, I'm not saying we should disregard scientific evidence, just that listening tests set a limit that might not be accurate. We're finding only things that can be tested, and many parts of the audio experience don't fit the test.

In this case, there are more bits down there, higher resolution at lower levels. It basically means you can turn up the amplification and receive more resolution regardless of amplified volume. But it also means you have more information, and that's worth something.

Dell probably didn't bother putting an analog reconstruction filter on their DACs, assuming they used an average Realtek codec the digital filter in the DAC has a minimum cutoff point around 28khz but could be higher when operating with higher sampling rates. They probably also sent the signal into a filterless class D amp to drive the output. Those two things together add up to a lot of ultrasonic crap in the signal.

By default, this is not true because sox will use a higher bit-count and perform dithering when converting to a lower bit format (ref http://blog.beatunes.com/2014/04/does-24-bit-audio-matter.ht... ) in order to remove the extra bits while reducing the impact on dynamic range.

In all fairness though, this is the same argument used in the Xiph article ( http://people.xiph.org/~xiphmont/demo/neil-young.html ) in an attempt to justify the use of 16-bit over 24-bit, the prior allowing for a comparable dynamic range with the use of appropriate dithering techniques. I just think it's important to note that the process used with sox is doing more than simple arithmetic bit-shifts/multiplication.

Anyway, nobody here is both qualified and willing at the same time to tell you what you need to know about this topic. The article referred to, written by Monty at Xiph, should give you a very good overview of how this works .

http://people.xiph.org/~xiphmont/demo/neil-young.html

Sampling theory says that a perfect square wave can be represented at any frequency below Nyquist. That doesn't mean that the codec or the analog electronics are capable of responding instantly at those frequencies, but that has nothing to do with the fact that a 1kHz square wave can be perfectly sampled with a 44.1kHz sample rate. The image is simply incorrect.

Transient response in the real world is generally limited only by the acoustic transducer response of the system, because everything else has the ability to respond much faster than audio rates. With Pono, this means that the earbuds or headphones you use with the player will have a greater affect on the transient response than the electronics inside.

21-bit would make for an awkward file format. If we can hear 17 bits of range that's enough to justify storing music as 24-bit.

I have very sensitive hearing, I hear things most others cannot. I can detect 192kbps vs 320kbps mp3 encoding. I cannot have any switch-mode AC/DC transformers in my bedroom because the switching noise actually keeps me awake (I charge my phone in the kitchen, most people think I'm mad when I complain about the "noise" :)

Still, I've never been able to detect frequencies above 20Khz or hear the difference between 16 and 24 bit. I think anyone claiming to is full of crap.

24-bit is pretty much useless for playback. While human hearing range may extend past 16-bits of dynamic range, that doesn't mean that full range is musically useful. It is, however, quite useful to record and mix in 24-bit for the headroom.

It looks like an 18khz sine wave, possibly with slightly reduced amplitude depending on the anti-aliasing filter rolloff and fc, but not enough to be audible (18khz isn't audible for a large portion of the population anyway).

> How is phase affecte?.

Probably delayed a bit by the anti-aliasing filter.

> Phase of an audio signal reaching the ears helps you perceive distance and position.

Not at 18khz, the wavelength is too short for your ears to notice any realistic group delay. High frequency localization is most done by ILD and effects caused by ear shape.

> What is the quantization error difference between 16 and 24.

Quantization error in a DAC just defines the noise floor. 16 and 24bit DACs are usually within a few dB of each other in dynamic range, it's really not audible.

> And if the author can't hear the difference between 4bit audio and 12bit audio I question what he/she is listening on. The aliasing would be HUGE.

What does aliasing have to do with bit depth?

but that doesn't have anything to do with number of bits? I mean, an audio player system is basically a digital stream sent into a DA converter outputting an analog signal which is then fed into an amplifier where the signal is amplified and then sent into speakers of some sort. No matter if you have an 8bit audio stream, or a 128bit audio stream, the stages afterwards play a much larger role in the hearing damage caused. Also not all DA converters ouput the same level to begin with. Some are -5V to 5V while others (single power supply) can be 0V to 5V, etc etc.

It can matter when you put it in a scanner though. I tried scanning some old photos, they look horrible compared to even my puny 8mp camera. Then I tried scanning a piece of cloth, and was blown away by the detail..