Is 24bit audio just a meme?

Most all equipment these days, especially low end stuff you neets jerk too.

16bit is enough for listening purposes.

But for processing 24bit lends better. Raising very low volume audio will help to bring out lost detail while with 16bit it will sound more muffled and generally have less effective volume range.

Total horseshit. Dynamic range doesn't change unless you are using digital volume control, which can take up to 12 bits away.

Most of the ADCs used in audio are of Delta-Sigma type. So they're basically 1 or 2 bit ADCs with a higher sampling frequency, they take the data and process/downsample it to 16-24 bits at 22 or 44kHz. If you check the datasheet of these ADCs you'll see that their noise floor usually corresponds to 16 bit, meaning that the last 8 bits are basically noise. But sometimes it is nice to have these 8 bits for cancellation and processing.

32 bit audio is literally a gimmick with no redeeming qualities that I can think of. There is no 32 bit capable ADC or even a resistor that has lower than 32 bit noise.

The only reason for 32bit audio is for extra headroom when processing and mixing. Absolutely no point in mastering stuff at that depth. Most recording equipment barely gives you an SNR above what 16bit offers.

While the range between the loudest sample and the most quiet will be likely the same in both cases (and you can imply it's dynamic range), however the effective detail between those two that can be manipulated will be lower with the 16bit recording than with 24bit.

32bit float effectively has no "ceiling" so it never clips. However that doesn't mean the playback system won't shit itself.

Usually the volume control can be implemented in the DAC + amplifier combo in two methods
Method 1: The volume control is implemented into the digital stage. The major drawback is that 1 bit of the D/A conversion is lost in every 6DB volume reduction. So, even though the built-in DA chip is 32 bit, in fact, only 16 bit may be left at low volume level.
Method 2: The volume control is implemented in the analog amplifier input stage through a volume pot to reduce the signal level. However, the volume pot affects the sound quality adversely. A low grade volume pot loses the details and creates channels imbalance resulting into soundstage distortion. Even a high grade volume pot inevitably loses the details. Both methods degrade the sound quality.
The best method applies the I/V conversion volume control, the volume control is just a variable passive I/V conversion being placed at the ACSS amplifier output, where the output is current signal but not voltage signal. I/V conversion is to change the volume level from the current (I) signal to the voltage (V) signal. (Like R-2R D/A chips output passive I/V conversion) It can keep the signal frequency band flat while not losing any detail. It does not degrade the sound quality in every volume level. There should be about four groups of diamond non-feedback buffer output stages that would offer very low output impedance.
Mark Levinson also knows that current volume control has great benefits, so in the volume control of his top grade hi-end preamplifier No.32, he uses many components to change the voltage (V) signal to current signal (I), then through the R-2R network to control the volume, and finally changes back to voltage signal (V) again. The best method simply works in current (I) signal domain, and technically, it is superior to the conventional technology due to less conversion.

The best method is to design digitally controlled relay-based volume control in 100 steps. It has 90DB control range ( most other volume control only had around 70DB range, so the device want to design the gain switches ) allow users to control the volume level smoothly. It should have two relay-based volume control channels through changing the DALE resistors to control the volume to avoid channel imbalance, achieving the best performance and sound quality of the gear. However, when you change the volume, the combine relays break or close, it may had slightly switch sound on output and disappear while stop change the volume.

You can add random bits to the end and get the same result. The datasheets tend to give the confident results, therefore some of the ADCs may actually give over 16b SNR but there's literally nothing out there that can offer 32b realistically. You can just fill out the rest of the bits yourself instead of forcing it on the ADC because it's not going to give you something that has correlation with the sound.

It doesn't magically increase the range of signal you can accept. As I said here you can just add the bits yourself instead of pushing the hardware to do it.

/thread

It's helpful in production. Unnecessary in playback.

What port is that connected to?

Someone post that screenshot of the forum where they are discussing the benefits of using different sata cables for audio quality.

you won't hear the difference on a speaker that shitty

it's a 2011-ish macbook/mbp so usb

i have a late 2011 with the same ports
pic related is a late 2011

Looks like the second USB port

Which raises the question of how good that amp can really be if it is only drawing from a USB port, even if the cables are short and heavily shielded.

Depends on the context.

24-bit audio for storing music is pointless.
24-bit audio for mixing can be beneficial (due to overhead).

But on the subject of 24-bit audio DACs, the thing to understand is that no 24-bit audio DAC exists. “24-bit” here means “above 16 bit”, and “16 bit” here means “below 16 bit”. So if you buy a 24-bit audio DAC, you can rest assured that you're actually getting like 18-20 bits of effective precision, which is way more than enough.

If you buy a 16-bit DAC, you're probably only getting like 14 bits, which is rather bad. That's why people go for 24-bit DACs, not because of 16-bit being insufficient, but because the ideal precision never being the actual precision.

16bit already gives you 96dB, which is enough.

I don't understand how somebody can *not* use digital volume control

Do you really want to be constantly reaching for your volume knob every time the track changes or you watch a video?

Lossless and uncompressed audio as a whole is a meme
> But user I can totally hear those 0's they add width to the soundstage and increase attack and speed in the bass (i roll 2d12+2 on dexterity and strength checks)
No
Fuck you
V0 is the perfect format, 320 MP3 is acceptable if you have autism.
Lossless is a meme and you shouldn't fall for it.

Most equipment can get past 20kHz actually. The issue, however, is that we can't hear whether or not it does.

24-bit performance however, has yet (to my knowledge) to be achieved.

Right, 96db is the range between silence and the sound of a jackhammer from a several feet away. 24-bit increases the range to 144db, which is the sound of a jet engine.
In order to even make use of 16-bit fully, your speakers would need to be able to accurately reproduce that range of sound without distortion, assuming the audio actually had a reason to produce such massive loudness swings.
Even very well mastered music usually sits at 20db encoded into the file, because otherwise soft sounds would get lost in relation to the loud ones.

That's why I was asking

>V0 is the perfect format
Spoken like a true tech illiterate

If you want to optimize for space, use Opus. If you wamt to archive or store music long-term, use FLAC.

MP3 has been deader than a brick since the last decade at least.

yes

Humans can hear up to 48Khz
16bit and 24bit can not be differentiated by human or machine

mp3 is just fine, 320 mp3 is as transparent as it gets

AAC is the best future format because its already in widespread use with it being the standard of audio streaming on the internet and having comparability on the same level as mp3.

There is no 'transparent as it gets' it either is or it isn't
Its not efficient at all for storage and it's not good at what you're using it for

FLAC takes more space, larger the file higher the risk of bit flip ruining the file

Opus is better than both MP3 and AAC even at lower bitrates. Please continue to explain to me why I should use a worse format that takes up more space for inferior quality.

Listening to music?
it sounds just fine

>FLAC takes more space, larger the file higher the risk of bit flip ruining the file
Why would bit flips ruin a file?

Ofcourse it sounds fine its TRANSPARENT you fucking retard it sounds the same as EVERY OTHER FILE
the only reason you use it is because you are retarded and don't use the smallest possible transparent file

OPUS barely edges out AAC while barely being supported

>barely being supported
It works fine on all devices I own including my PC, my netbook, my phone and my tablet. I don't know what else I'd possibly need support for Opus on.

>OPUS barely edges out AAC
Opus is also free whereas AAC is proprietary

Its supported if you don't use garbage software and hardware
Its still the best format

flac 16 bit/48kHz is fine

>le proprietary meem

AAC is supported on every music player past 2005, while opus isnt.

apple doesnt support opus, but its hard not to imagine why

Opus is supported on the good ones
Why would you use something shitty from apple

I'm unable to tell the difference between above 160kbps mp3s, does this concern me?

>apple doesnt support opus, but its hard not to imagine why
Apple doesn't design good hard- or software?

Apart from the fact that you could make your music library half as small by using 96 kbps Opus instead of 160 kbps MP3, the bit depth of the DAC is mostly a function of how low the noise floor is.

You don't need to be able to hear distortion in order to be able to hear noise. Even a shitty codec can have a low noise floor.

This post makes no sense. Why would you have to reach for the knob in an analog setup?

Analog volume control requires an analog input

not really, just means you can't hear much above 16khz which is the case for most people by a certain age.

It also means he can't hear whatever distortion is caused by 160 kbps MP3.

Contrary to common belief, the frequency cut-off is not the only thing that distinguishes lossy codecs.

Haha no

Now I'm confused. in b4 something retarded like software controlling motors on his analog mixer

Either your volume is controlled inside your dac,(digital) or elsewhere in the signal chain.

The question was how to *not* use digital volume control

You can't answer “inside your DAC (digital)” to “how to *not* use digital volume control”

There's literally no reason to use lower quality audio over the higher quality audio, if you have it. It doesn't even matter whether you hear the difference.

I use 24/96 for recording, mixing, mastering and playing live. Also for archival of my own stuff.

For listening at home it's completely unnecessary.

>Lower storage cost
But feel free to encode everything in uncompressed 32-bit 192 kHz PCM if you have more money than brains, I suppose.

Thanks, I already keep DSDs on my phone.
>storage cost
Are we in the 90s or something. There are 16TB SSDs out there. Like, come on.

>Are we in the 90s or something. There are 16TB SSDs out there. Like, come on.
And they cost several thousand euros.

If you have the ability to piss away several thousand euros on something you're literally incapable of hearing, then you might as well give the money to somebody who will actually do something useful with it instead

On playback high samplerates sound worse

That does not actually contradict his point. If 44.1 kHz is higher quality than 96 kHz, then as said, there's literally no reason to use 96 kHz when a 44.1 kHz version is available.

If your speakers are shitting themselves from the HF, sure, very possible.

>56748855
>Humans can hear up to 48Khz
lol

>Humans can hear up to 48Khz

No.

holy crap
people talking out of their asses - the official thread

Oh and I think it's suggested to record in 24-bit for producing so you have more room to work without much distortion or something.

Sample rate not frequency retard

Both trivially false

1. Human hearing in a child maxes out at around 20 kHz. For adults it's even lower.

2. While it's not possible to hear a noise floor of -96 dB, it's very easily measurable

20 kHz frequency = 40 kHz sampling rate

Adult human hearing maxes out well below that

You really don't understand, do you?
Digital signal is sent to a DAC to convert it to analogue signal (voltage or current) that can then be amplified. Either the volume control is handled digitally inside your dac, or somewhere after the dac, raising or lowering the voltage. (Analog)

Do you believe in digital amplifiers? How can you amplify ones and zeros?

The question was how to *not* use digital volume control. Emphasis on *NOT*

This is my last reply to your shitty bait. You can either explain what you meant by or fuck off.

I wrote a very large two part post earlier in the thread, fartsniffer.

It contains your questions, answered hours ago.

Easy answer would be a volume pot, but those have all sorts of problems also.

This is the post, google

the human ear can't hear past 128k

Well fartsniffer?
"Won't reply because I have no clue how DACs work?"

This still does not answer my question. I'm beginning to think you simply don't understand me. Let me try rephrasing by using a hypothetical thought experiment or three:

1. You play a youtube video, and it's too loud. Do you

A) Adjust a physical knob to make it less loud
B) Electronically control the output gain of your DAC (e.g. alsamixer)
C) Use software volume controls (e.g. key or slider on the UI of the program)

2. You want to play a mixed set of musical tracks, but they're all mastered at different volume levels. Do you

A) Adjust a physical knob every time the track changes
B) Electronically control the output gain of your DAC
C) Use software volume controls manually (e.g. key or slider on the UI)
D) Use ReplayGain tagging to apply a precalculated, clip-free gain level for every track (thus making them the same loudness)

3. You want to play a simple video game, but the video game's rather basic sound effects are uncomfortably loud. Do you

A) Adjust a physical knob to make the video game less loud
B) Electronically control the output gain of your DAC
C) Use software volume controls (e.g. option in the video game's settings menu)

?

>1.close the video
>2.delete the louder track
>3.puncture my eardrums
Never though Cred Forums was really so detached from the real world.

>How can you amplify ones and zeros?
nice bait

humans don't 'hear' sample rates you retard.

Again, I do not give a shit about how your DAC is built.

Let's settle a few assumptions. Tell me which of these is wrong in your world view:

1. Your volume adjustments are either digital or analog.
2. Once the output signal crosses over from digital to analog, it doesn't go back
3. Per-stream volume adjustments must be done before mixing
4. Stream mixing is done digitally

From these four assumptions, which are all true in any system I know, we can conclude that:

1. Analog volume adjustments, will affect all streams equally

2. To get per-stream adjustments, you need to use digital volume controls

In other words: If you're switching between sources (e.g. different tracks, different games, different videos, whatever), you either

A) have to constantly adjust the analog volume level (e.g. knob)
or
B) have to use digital volume controls

For most people, constantly fiddling with the volume knob is annoying, therefore we use digital volume controls. Since you look down on digital volume controls, that implies you enjoy fiddling the volume knob constantly, hence my question in QED

Question: are you using the optical output of your motherboard, or the analog outputs (headphones or external amp)?

If it's the optical (digital) it's up to the designer of whatever DAC you use to decide the volume control selection. Most all will use digital volume control, implemented inside of the DAC.

If you are using the analog outputs, you have a DAC/amp combo onboard, which most likely uses digital volume control.

It is always the best choice to leave the volume of your dac at full output, to preserve the full bitrange. The volume can be controlled with a pre-amp that way, preserving all your precious bits.

All your options assume your computer is controlling the volume, which almost guarantees digital volume control.

Ever wonder why your music sounds "better" when you turn it up a bit? That's you hearing more dynamic range lost through volume digital volume control.

YOUR "ANALOG" KNOB IS JUST A POT THAT GOES TO INDIVIDUAL PINS ON A DAC.
just because it's a knob does NOT mean it's analog.

Go look at the white paper of a sabre DAC(or whatever) and look at the pin out if you don't believe me.

>Question: are you using the optical output of your motherboard, or the analog outputs (headphones or external amp)?
Answer: Neither. My DAC is connected via USB. But for the sake of your question, this is effectively the same as what you call “optical output” (i.e. digital). (And I'll give you the benefit of the doubt here and assume you were also aware that you can pass S/PDIF via RCA connections, which is more common than optical anyway)

>If it's the optical (digital) it's up to the designer of whatever DAC you use to decide the volume control selection. Most all will use digital volume control, implemented inside of the DAC.
Actually, it isn't. I'm fully in control of volume control, because it's done before the signal even leaves my PC. I do it using PulseAudio, which allows me to adjust the volume of streams individually. I have streams default to 70% volume (-10 dB), which gives me more than enough headroom to make too-quiet stuff louder without clipping.

>It is always the best choice to leave the volume of your dac at full output, to preserve the full bitrange. The volume can be controlled with a pre-amp that way, preserving all your precious bits.
I do leave the volume of my DAC at full output (well, actually I leave it at -0.5 dB, but that's besides the point), but not for the reason you seem to be suggesting. I do it because again, I do volume controls with PulseAudio, rather than in hardware. This allows me to do it on a per-stream basis.

>All your options assume your computer is controlling the volume, which almost guarantees digital volume control.
Good job picking up on that sherlock.

>YOUR "ANALOG" KNOB IS JUST A POT THAT GOES TO INDIVIDUAL PINS ON A DAC.
Trivially false. My analog knob is built into my amplifier unit, which has absolutely nothing to do with my DAC. My DAC doesn't even have a volume knob. It's a PCB. (ODAC revB)

I'm beginning to think you have absolutely no idea what you're talking about.

potentiometers usually are analog. people wouldn't use a potentiometer with discrete resistance in this case.

Not sure about that. Cause I can hear audio much more clear on 24bit then on 16bit. but may not be noticeable to most people.

44.1khz and 48khz I can barely tell though.

Same wrt 44.1 and 48, but 96 sometimes gives me headaches

The spdif is not your analog output, it's still digital. that is not what I was asking. You use digital output to your dac, then?
An amp has no volume control, so yours has a preamp built in, obviously. Give me the model and u will tell you if its analogue vs digital.

>Cause I can hear audio much more clear on 24bit then on 16bit.
bullshit bullshit bullshit bullshit bullshit

Okay, at this point I'm 100% sure you're intentionally trolling me. Your responses are incoherent, and it's like you're not even reading what I'm saying in the slightest bit.

This is my last reply, have a shitty day

Yes, thank you for that useless trivia. The pot controls the DAC. Nothing more. The magic happens inside. Go look at a pin out for God's sake. It's a loop.

if the analog potentiometer controls the amplitude of the output of the dac, then it clearly is analog control of volume.

Your computer is controlling the bit depth. It's fucking digital, retard, all software controlled. Your computer is not outputting all 24 bits or whatever when you turn the volume down through software. Understand now?
Control the volume at the hardware level if you want the full bit depth, even then it's iffy if you are getting full bit depth due to the designers whim/budget.

Nope it's digital inside the DAC. I'm too lazy to find the white paper of a dac, so go find one, and learn something. Analog volume control happens to the either voltage, or the current output of the DAC. The output. After the DAC.

Digital volume control: at full power a DAC might put out 1.5v or something. If it's digitally controlled, it would be less, say 1.25v or something.

Analog happens after the DAC to that 1.5v signal or whatever the DAC is outputting.

there's nothing digital inside a dac. only the inputs are digital. have you never seen a dac datasheet?

Go look up ess sabre 9023 DAC and then tell me the volume control is not digital.

Retard

stop responding to a shitty troll, please. It hurts to read

In the Sabre DAC the volume adjustment is done prior to the Oversample Filter (which is also before the built-in ASRC).

In another dac, the Wolfson 8741 which has an internal path of 24-bit has dithering control in the register setting. The datasheet specifies:

"Dither is applied whenever internal truncation occurs. It is also used when 32 bit input word is applied to the DAC prior to truncation to the internal wordlength. Three types of dither can be selected (in the digital filter) to allow the sound quality of the device to be optimized."

Compare this with the “traditional”(analog)way of handling volume control: pass the signal at full magnitude through the DAC and then do attenuation in the analog domain through a preamp. If over aggressive signal peaks generate distortion in the DAC, they these can only be attenuated in the preamp but never eliminated.

>I have no arguments left

You sent getting all 16 bits if you use digital volume control. That's why 24 sounds better to him. Digital volume control sacrifices bit depth, reducing quality.

>not wanting to turn the knob on your beautiful McIntosh C32

16 bits is enough even with digital volume control

For you maybe. Reducing volume in software is basically equivalent to reducing the bit depth. In digital audio, the signal is split up into distinct samples (taken thousands of times per second), and bit depth is the number of bits that are used to describe each sample. Attenuating a signal is done by multiplying each sample by a number less than one, with the result being that you're no longer using the full resolution to describe the audio, resulting in reduced dynamic range and signal-to-noise ratio. Specifically, every 6 dB of attenuation is equivalent to reducing the bit depth by one. If you started with, say, 16-bit audio (standard for audio CDs) and reduced the volume by 12 dB, you'd effectively be listening to 14-bit audio instead. Turn the volume down too much and quality will start to suffer noticeably.

Another issue is that these calculations will often result in rounding errors, due to the original value of the sample not being a multiple of the factor by which you're dividing the samples. This further degrades the audio quality by introducing what's basically quantisation noise. Again, this mostly happens at lower volume levels. Different programs might use slightly different algorithms for attenuating the signal and resolving those rounding errors, which means there might be some difference in the resulting audible signal between, say, an audio player and the OS, but that doesn't change the fact that in all cases you're still reducing bit depth and essentially wasting a portion of the bandwidth on transmitting zeroes instead of useful information. I made a post in which I explained how digital volume control can be implemented without the loss of bit depth.

that sabre dac has more stuff inside it than just a dac.
look at dac0800 for comparison. the dac alone just converts the digital value to analog. there is no reason to have anything digital inside it.

>For you maybe.
You can't hear a signal at -96 dBFS. You are full of shit.

Makes no difference because the volume control is just registry values. That's it.

I am so sorry you can't comprehend digital volume control sacrifices bit depth. Where are you getting the -96db from?

>Makes no difference
then you agree that volume control with a dac is analog.

16 bit = 96 dB range. -96 dBFS is so far beyond the range of hearing it's silly.

Heck, even if set your volume to 25% (-35 dB) you are still getting a noise floor of -60 dBFS compared to your signal. This is what it sounds like: 0x0.st/QqC.flac

If you claim you can hear that underneath a signal (without turning up your amplifier), you are full of shit.

Nope it's not. 100% do not agree.

Its Done digitally inside the DAC through registry values, essentially.

I already explained analog volume control here

Human ear has about 120db, buddy. More like 130 if you are into pain.

>missing the point
You are absolutely retarded if you honestly music is mastered with 120 dB of dynamic range.

120 dB of dynamic range is the difference between the eardrum-blowing pain of a jackhammer at close distance and the extremely engineered silence of a completely isolated anechoic chamber.

You do NOT need that amount of dynamic range. 24 bit precision is a SCAM.

everything you say in is about signal processing, not the dac at all. read this again .

If you have to ask then the answer is yes.

A great way to see if your system has full range is the1812 overture. The cannons, if mastered correctly will have 120db of range. Oldest sound test in the book. Californication, or some other shitty recording sure won't have 120db of range

Lol What do you think a DAC does, again?

How about you explain it to me since you know so much. All I keep reading is "no it's not" or " your wrong and full of shit" or something like that. I have taken you pretty much through the entire process with my posts. Now it's your turn to describe in detail how a DAC and digital and analog volume control works.

>Is about signal processing, not the DAC at all

I can't stop laughing at this, thank you

Getting arrested for playing 120db fucking audio

*Dynamic range*

OK I'll let you in on a secret if you won't tell anyone: the ess chip has an onboard DSP also! Hush hush! It's how digital volume control works! Tee hee!

You fail to realize that you aren't getting 24 bits of precision if you are using digital volume control.

>Do you believe in digital amplifiers?
I believe they're called class D

checkmate atheists

What do you think dolby and DTS are, and what do they do?

>What does a DAC do?
converts digital values to analog values. as simple as that. that's why it is called "digital /analog converter". it's not a hard concept if you think about it.

It's class D. The "digital" part is just a buzzword to sell stuff to clueless idiots.

Now please explain how that you would amplify a one or zero.

>A great way to see if your system has full range is the1812 overture. The cannons, if mastered correctly will have 120db of range. Oldest sound test in the book.
Holy shit, audiophiles are hilarious

Is this the new /audiophile/ thread?

thanks for telling me your deep secrets, I guess

But most dacs also have a DSP built in to decode the format. Again: what do you think Dolby and DTS are, and what do they do?
Does the music on your CD /mp3 just come into your dac as a random stream of one's and zeros? What order, and how long are the "words" decided?

then maybe this is the contraption you're trying to describe kind user newegg.com/Product/Product.aspx?Item=9SIA7BF2K21044&nm_mc=KNC-GoogleMKP-PC&cm_mmc=KNC-GoogleMKP-PC-_-pla-_-Eco Gadgets-_-9SIA7BF2K21044&gclid=Cj0KEQjwpZO_BRDym6K_nMye7cEBEiQAVA7RaEvX6b1EqhiAkFbEKdsk2MSlZT6e3i4PjKjRhm0TlvMaAp4F8P8HAQ&gclsrc=aw.ds

I'm guessing that your system does not have 120db of range. I'm sorry if the truth is different from your preconceived notions of what is good enough for you.

I mean, is that all you can respond with? Nothing else other than to attack an argument with scorn and derision, without any fact or reason? " Holy shit /g is so retarded but I can't tell you why because I don't know and it would mean changing what I thought I knew before so I will just meme away."

The volume control to the WM8741 DAC for instance, is implemented in the i2c bus. the uC talks to the DAC through the i2c bus and it also talks to the LCD through the i2c. You will need a 5v to 3.3v level converter because the arduino uses 5v signals and the DAC works with 3.3v signals. That's what that's for.

>But most dacs also have a DSP built in to decode the format.

>dsp+dac
>call it a dac
nice

Poorfags, poorfags everywhere.

I don't make the products or name them.

>CPU has CPU+GPU built in
>Call it a cpu
Nice

ARE DAC REALLY NEEDED?

ANSWER THIS QUESTION:

DO YOU HEAR STATIC BACKGROUND NOISE OR INTERFERENCE?

IF YES, BUY A DAC. IF NO, FUCK OFF

If you want to hear sound playing, then yes. Chances are your computer or phone already has one inside. Do you like the way your device sounds? No? Then maybe the $3 DAC inside is not good enough for your tastes, and you may need to purchase an external unit with higher quality components and better design.

Digital audio is all a FUCKING MEME, enjoy your aliasing, ADC and DAC noise.

Why do audiautist need expensive gear to "listen" to music when there is almost no difference with reasonnably priced gear?

In your case, you're implying you're going to mess with a digital volume control or an analog one. So if you're not doing it with an analog volume control, you'll do it with the digital. I've always found digital controls a little more finicky than digital ones, and analog ones are easier to adjust. The fact that changing the analog one seems to be a pain, might just have to do with your setup and not analog volume controls themselves.

>expensive gear
same applies to fucking photographers and $10,000 gear faggotory

Except spending $10k gear doesn't mean you're going to magically take good photographs.

Sound engineer here, I usually work in 48k or even 88,2K to get the 44K1 lacking ultra highs back. 24bit is good for recording, overkill but better than 16bits if you failed your recording and have to get a lot out of your audio clip. For playback >48K is useless.

Forgot to mention 16 bit is way enough for playback as well.

''Is x a meme ? xD''

You're literally the worst cancer straight from Reddit.

Kill yourself for asking this question.

For source material that you're only going to listen to, not edit? Absolutely.

For DAC? No, that can be useful.

>But most dacs also have a DSP built in to decode the format
oh, so this is what all your misinfo is based on.

no, that is not a DAC, a DAC may be part of that system, but in it's entirety it cannot be called a DAC. the DACs people into music use are d/a converters, nothing more. they have no volume control, not format decoding. the volume is controlled in software by the player and/or other software it's connected to, and/or by the amp which is entirely separate and a completely analog device.

you're thinking of home theater shit or placebo formats for idiots who throw money at stuff that doesn't sound any better.

Some people care about the expensive price, it's can be a reason for purchasing with the exclusivity and rarity that often comes with. People interested in getting a good sound might not have to spend much at all but if you want the best, you are going to have to pay. The one thing many still don't seem to get is how price has next no correlation with sound quality. While the best gear costs a lot, there is a mountain of terrible and expensive gear on the market as well as a lot of good and cheap stuff. Achieving good sound in a room goes beyond getting good gear.

Poor analogy. For a professional photographer it is a tool and an investment. It is a a necessity to get the job done and camera/lens market is much more competitive and less skewed than consumer audio is. Aside from a few select brands like Leica who focus a lot on the branding and status, paying more often gives you obvious returns. While even modest DSLR/mirrorless system gives you the image quality needed, that's not what really differentiates the professional camera gear from things aimed more for hobbyists and amateurs. You pay for robustness, ease of operation, better battery life, better focusing system and above all, something that is reliable and allows for a fluid work flow. 10k for a hobbyist is quite a bit but isn't for a professional and no amount of money makes you take great pictures. Besides, a photographer either way is an artist here and not the consumer unlike a person who listens to music.

A better analogy would be some connoisseur who likes to look at photographs/videos and spends a lot on the system to display them. I think there's nothing comparable to the nutjob "high end" audio enthusiasts though.

How does 48k improve over 44.1k strictly for listening purposes?

Makes it easier(cheaper and more effective) to do a low pass filter in the DA-converter before Nyquist without altering frequencies in the audible band(below 20KHz). It doesn't improve anything during listening on itself as 44.1KHz already covers all audible frequencies.

I didn't read all the posts in this thread but for the most part I've read many people saying adjusting digital volume kills the bit depth, which is true.

BUT

I just wanted to add that afaik this is not a problem in Windows since Win Vista (inb4 winfag), because everything gets upscaled to 32bit first, then yes, digital volume adjusting can occur so that the bitdepth lowers a little bit (pun not intended hehe, anyway -1 bits every 6dB), BUT in the end the final bitdepth gets rescaled to 16 or 24 bits according to the file. Of course this leads to some quantization errors but everything it's better than going down 16bits.

24 bit is cute but vinyl has INFINITE fidelity because it's analog
digital = literally hitler

Incorrect on all points. Congratulations for being the stupidest poster in the while thread.

It's a meme, just like 10-bit displays.

HDR displays aren't a meme.
It's interesting how audio technology advanced sufficiently to achieve complete transparency to the human ear more than 30 years ago, but display technology still has a long way to go.

I actually find it baffling that we've gone so long using displays with 8 bits per channel or worse, when the stair steps are clearly visible. It's about time this was improved.

So wait a minute.... 32 bit DACs have a purpose, then... so high bit is there for a reason then? It's not just a meme?

If you have the right equipment and the right music, yes, you can live the "32 bits experience!!1!" (with absolutely 0 difference compared to a true 16/44.1khz listening).

It's funny because those signals get upsampled to 24 or 32 inside a DAC to overcome the inherent disadvantages of low bit depth signal processing. It's pretty obvious why an OS would oversample your 16 bit stream to 32 for most people, but you on the otherhand....

>It's pretty obvious why an OS would oversample your 16 bit stream to 32 for most people
That's the point of my first post, you dense, so you know that I know. Just to explain some others tards above.

What the fuck user. Highest dynamic range within a recording I've seen is just inching above 20 dB. That's getting to the silly territory where if you want to listen to it properly to hear the quietest sections well the amplitude peaks are 20 dB above it and start to get really fucking loud. Some movies go near 30 dB and the effect is even more pronounced. Speech on a comfortable level means action scenes are very loud. If you have a system capable of playing above 110 dB continuously, it's actually quite fun when shit gets loud when using all of the power your amplifiers can deliver and loudspeakers can handle... and your ears.

There is no recording which has anywhere near 16bit dynamic range. The formats may have a capability of storing such precision but the recordings do not nor should they. It would be impossible to listen and there are no such differences in any instruments or voices. 24 bit precision exceeds the capabilities of any DA-converter or amplifier on the market anyway. The fact that they can handle such bit depths as a format doesn't mean that they output such resolution. Noise floor on the best convertes is at 21 bits.

Digital volume control takes one bit away for every -6 dB. Turn the volume of down, you sent playing with 16 bits anymore. Understand???

I perfectly understand how reducing volume digitally creeps up the noise floor. For it to become audible however you need to go crazy with even 16 bit resolution. The noise floor in recordings is orders of magnitude higher than what there common formats limit. Dithering and noise shaping give even 16 bit resolution a rather huge headroom of adjustment. And just to be clear, I don't oppose the higher bit depths at all. There is no downside and I've set my output format to be 24bit, highest my DA-converter can handle. My last post was against the ridiculous notion of some tracks having 120dB dynamic range.

A DAC will typically have 80-95 dB of range.
An orchestra is recorded while playing a symphony and the dynamic range of the performance is a maximum of 65 db above the noise floor - so if the noise floor is at 30db then the maximum dynamic range required to record the performance without distortion at the peak volume would be 95 db (30db + 65 db).

The maximum dynamic range of a 16 bit file is 96 db and that of 24 bit is 144 db

So the dynamic range of the original performance was 65 dB, then digitizing it to more than 65 dB is a waste.
If a symphony was limited to 65 dB dynamic range, then 11 bits would be enough.
But is 65 dB a realistic number for a symphony performance? Does that allow for the very quietest moment (when no one is playing, speaking, or coughing) to the very loudest drum crash or cannon (think 1812 Overture)?
The dynamic range of human hearing is often estimated at 140 dB (threshold of hearing to threshold of pain). I don't know how much a symphony performance could get to, but 65 dB seems pessimistic.

I build speakers and amplifiers for a living let me explain some things.

1. 24 bit is best for mastering where the overhead is needed. 16 bit is the actual limit of transparency. 320kb/s is not actually transparent and like any lossy format is full of artifacts. Get the CD or at least a high quality digital file if you care about accuracy in your music playback.
2. Most audio products are really shitty and have poor accuracy as far as frequency response. High order crossovers are a compromise for poor driver pairing in loudspeakers. Consider 1st and 2nd order crossovers when buying passive loudspeakers. If going active make sure each driver has its own individual amplifier.
3. Class of amplifier makes a huge difference in sound. Class A and class A/B generate a full waveform and are superior to class D designs at all volumes. Class D doesn't even generate a waveform (your pulse width modulated class D desktop amplifier is not high fidelity and full of distortion).
4. Analog will always sound better than digital because it's impossible to generate a linear waveform through digital means. Yes vinyl and another analog means of audio production will inherently sound better if you invest enough into proper playback.
4. Time domain transforms are one of the most audible aspects of music reproduction. Major timing delays between drivers ruin stereo imaging. Stereo imaging is essential for playback realism.
5. Vacuum tube amplification and properly designed non digital solid state amplification produces a more accurate waveform with less time domain transforms. The less negative feedback the better it will sound. Negative feedback helps with total harmonic distortion, but you remove depth and it usually makes things sound "brittle."
6. Dipole loudspeakers designs offer the best sound possible. They are usually inherently time aligned and the comb filtering from the backwave creates a more stable stereo image from the listening position.

Feel free to ask any questions.

The other thing to keep in mind is that you would need to leave some headroom at the high end unless you knew in advance what the highest peak level of the performance was going to be. If you guess too low, you will clip on the peak. If you guess too high, you will be 'throwing away' that much dynamic range. So you might want 10 or 20 dB extra just to be sure you stay out of clipping. Then your 65 dB symphony would need to be recorded with 75 or 85 dB of dynamic range available.
I don't think anyone worth a shit would argue against recording at 24 bits. Apart from the headroom issue, it also allows lots of editing and DSP to be done without accumulating quantisation noise. But once it's all done and dusted, there's no point packaging it for distribution at anything more than 16 bit.
Apart from all this you are confusing a 24bit recording with a 24 bit DAC. There's a damn good reason 24/32 bit DACs are a thing, and it primarily has to do with:

Digital volume control

...

Stupid fucking shit has never heard of hypex clasd d amps, have you?

Go die in a fire

My anaconda swallowing my speaker cables really improved my audio setup.
Weird how an cold-blooded animal makes the sound so much warmer.

They're okay as long as you keep the volume low. Technically they operate in class A/B as long as you don't push them too hard. I was referring to stuff like lepai amps and nuforce stuff.

Lmfao I literally said nothing about cables. They barely matter, as long as you aren't using the shittiest of shit. Silver may sound a bit brighter but most things are audibly pointless past well made copper at the proper gauge for your running lengths.

Idk what you just said but riddle me this, do Flacs sound better than mp3s or is it just my placebo?

hypex.nl/component/weblinks/weblink/24-datasheets/28-nc400-datasheet.

Please explain how this class d amp operates a a/b.

Yes they do. They are free of artifacts and have no issues with dynamic range. It is not placebo, lossy audio is a compromised system of playback.

depends on the flac and the mp3

and it depends on your audio output equipment

Doesn't matter too much if your os/software up samples everything to 32 bit anyway.

I'm using a fiio m3 with a pair of Sennheiser earbuds, as compared to my phone it seems to me it sounds better but then I see all these people on the internet telling me my ears shouldn't actually hear the difference.

how do you feel about the powered speaker meming on this shitty board?

how do you find the JBL LSR 305

It's using MOSFETs that are fed into a digital correction chip. At lower wattage the correction chip doesn't even turn on, thus making it class A/B at lower wattage.

im not the hi-end audio dude, but i have always been able to hear differences between poorly encoded 128kps and well encoded 128kps mp3;

a few well encoded 128kps mp3 are almost as good as well made wavs and flaccs, as well as CD tracks, except for certain passages where the treble can be indistinct or the bass is absent or boomy or the midrange is lacking full detail

i usually can't tell much difference between 320kps mp3 and better made wavs or CD tracks

i usually use amped HD650s, or Quad 21 speakers running off dual mono amp SAE or Cambridge Audio class A amp

Who encodes with anything other than LAME

i use wavelab

It turns out it has a tiny class a amp, boosted by class d, making it technically a/d. But it's a 400 watt amp, and the first few watts being class a are somewhat irrelevant. And I'm looking at the graphs trying to figure out where this supposed drastic change in fidelity happens and am not finding it. Some help, maybe?

For the price they're good, but if you want accuracy consider genelec, emotiva, higher end JBL or similar. As far as active design the king Roy kit from madisound is top dog for full range performance and stereo imaging accuracy.

Do you really think the psycho-acoustic modeling behind audio compression is perfect? It comes close, but it's not. That's why lossless is better, even if it's just 5% better.

interesting
thanks

The first few watts being class A are highly relevant if you're using efficient enough transducers. The major change in fidelity is due to the simple fact that class A and class A/B can generate a 360° waveform and 180° waveform.

Time to get better amplifiers and speakers. Most of the distortion is there, not on the signal source.

I understand and agree with you, but I'm just failing to see where, precisely, the fidelity difference occurs. In the end, when you have the volume up to a moderate level, you are still essentially listening to "mostly" class d?

They are different algorithms (modelling+ encoding) to do lossy audio compression. Dolby AC3 is related to AAC and DTS is related do aptX. What did you mean, user?

Stop confusing dynamic range with max SPL. Your amps will have 25dB of noise floor at least, your living room will have 30-35dB of bg noise. To hear 120dB over that your system without have to be pulling 150dB SPL and you'd bleed out of your ears.

In a concert setting, a fortissimo symphonic orchestra will pull around 110dB SPL peak (95 RMS) and your cannons will be 12 dB over that, or the audience would have to wear earplugs.

Mechanical inertia in the recording lathe and in the playback cartridge makes infinite bandwidth impossible.

Focal Be6 twin

MK 1611p are also nice.

Integrity check nigger
Flac has it

there is more than 1 DAC topology available, you fucking nitwit

>how you would amplify a one or a zero
those ones and zeros, when combined together, represent numbers
you can multiply numbers by other numbers
the result is then converted into an analog signal, the higher the numbers, the higher the output voltage/current

You replied to a bait but same goes for everything analog domain. The size of an atom limits vinyl's absolute theoretical resolution far below what is already in use in digital formats today.

Always found the bass to be smoother in 24 bit files and it sounds clearer.

Maybe I'm just imagining it but it sounds better to me so I'll carry on liking it.

>do Flacs sound better than mp3s or is it just my placebo?
Could be either. Too many things go undefined here. To test it, you HAVE TO double blind the test.

holy shit
if a component costs 3 bucks when you are mass producing computer motherboards, it sure as hell is going to sound absolutely fucking fantastic
in the real world, electronic components are much much cheaper

>thinking a DAC is a magic expensive box that audiofools buy for their expensive snakeoil headphones
jesus fucking christ, educate yourself, man

There is a hidden assumption you are making with dynamic range, that the spectrum is necessarily flat. It is common procedure to manipulate the error spectrum to more closely align with MAF, the hearing threshold.

>16 bit is the actual limit of transparency.
Not really, no.
>High order crossovers are a compromise for poor driver pairing in loudspeakers
They serve the additional purpose of limiting radiation pattern interfering between the drivers, for the purpose of matching directivity.
>Class D doesn't even generate a waveform (your pulse width modulated class D desktop amplifier is not high fidelity and full of distortion).
It does generate a waveform. This is then fed into the RC filter to remove high frequency junk.
>Analog will always sound better than digital because it's impossible to generate a linear waveform through digital means.
The point is to have a filter remove the aliases to get proper reconstruction, as per the sampling theorem. Oversampling helps. So no.
>Time domain transforms are one of the most audible aspects of music reproduction
At the scale of errors committed typical of loudspeakers? No.
>Vacuum tube amplification and properly designed non digital solid state amplification produces a more accurate waveform with less time domain transforms
>The less negative feedback the better it will sound. Negative feedback helps with total harmonic distortion, but you remove depth and it usually makes things sound "brittle."
Just no.

people.xiph.org/~xiphmont/demo/neil-young.html#toc_wd2bm
It only matters for audio recording. 16bit is fine for audio playback.

Lol thanks for that. I'm not the one confused
Lol no. Signal path design, component choice are vitally important also. That $3 chip might be awesome, but cheap Chinese resistors instead of DALE, or a shitty power supply makes ALL the difference.

>56760364 Silver may sound a bit brighter

You may delete yourself now

design is obviously going to be the most important aspect
the choice of resistors is barely going to make a difference, at least in audio quality, let's leave reliability aside
making a good quality power supply is absolutely trivial and i'd be surprised if people using 3 dollar components in their circuits managed to fuck that up

The limit of transparency is a moving target dependent on multiple factors. 16 bits could in some extreme cases become audible but mostly that's quite easily in the transparent domain. Even 8 bit resolution can be enough for extremely low dynamic range pop music as auditory masking makes sure you will never hear the noise floor through all the loud sounds saturating the audio band.

Bitrate is a poor indication of anything. 320kbps mp3 with a modern encoder is almost certainly transparent aside from problem samples where the(encoder dependent) quantizer fails. Encoder alone can make a massive difference in quality. Grab an old mp3 encoder from the 90s and encode a song at CBR 320 kbps. It's likely going to have some audible artifacts. Then use the newest LAME and encode with variable bitrateat the highest setting(avg around 250kbps) and I guarantee you will have trouble hearing any kind of difference between lossless and the lossy encode. Codec listening is boring crap anyways and above all a learned trait, not some magical limitation of your equipment like many people like to think it. While lossy encoding obviously throws away information, it doesn't mean that this is audible even under very strict test conditions.
>I actually work in the high end audio industry.
I believe you.

Oh hey it's the nerd. Forgot to thank you for your efforts at... "jargon" in /hpg/. Thank you. Made me start Googling to understand things further but the more I research the more I think I need to go back to basics and grab a physics book or two.

>They serve the additional purpose of limiting radiation pattern interfering between the drivers, for the purpose of matching directivity.
Sounds like something most manufacturers really wouldn't bother with. Also as far as I understand, I wouldn't consider crossovers as a compromise for poor driver pairing as you could never match drivers as effectively as you can with proper crossover design, especially active ones.

Nothing in the Hypex documentation suggests a second amplifier built into each module. Class D does not switch off the modulators at lower power modes, and the amplifier efficiency in low power use suffers.

>Silver may sound a bit brighter
>I actually work in the high end audio industry
Yeah it's basically confirmed. Is gold warm yes?

Yes it is. Some engineer put 24 bit matrixed with a 16 bit. Then they played the music (24bit one - 16bit one): Zero sound, only one or two clip and or inaudible artefacts at +120dB.

4. Analog will always sound better than digital because it's impossible to generate a linear waveform through digital means. Yes vinyl and another analog means of audio production will inherently sound better if you invest enough into proper playback.
Stopped reading there. Bullshit.

I like cute neets like you who just blatantly ignore facts on simple objectivist assumptions without looking more into the subject.

I'm an IT professional who has studied digital and analog signals as part of my degree.
Do you understand how sampling theory works?

>I'm an IT professional

Gold is used because it doesn't corrode. Again, when silver wire is used nothing changes that's measurable, but some people perceive the sound is brighter.

Again, high end cables are bullshit, you just don't want to use the cheapest shit possible that may have conductivity/reliability issues or if the gauge is too thin for your running leads.

>fedora because I have no rebuttal
Oh, and let's not even get into the whole fucking story of how most vinyl released in the last 40 years has already been pre-digitized.

You argument is fucking invalid.

I'm genuinely curious how you achieve a selective understanding of some field like this. You know some things and are blatantly wrong on others to the point you are parroting audio myths. Salesman?

Alright, I admit, I was just trolling.

Lmfao this isn't even me. Some samefag is butthurt I guess.

Im a designer. I was taught by Nelson Pass. Most engineers have esoteric beliefs about certain technologies and it's pretty common outside just high end audio and stereo. Not everything is digital.

>I actually work in the snakeoil sales industry
consider killing yourself

I'm kidding of course, it was me all along. I'm an interminable faggot and I just cannot stop sucking dicks.

No thanks, I have a wife and kids and make a lot of money. Considering your edgy habits maybe you should consider such actions.

6moons.com/industryfeatures/ncore/1.html

Read down a bit about the class a part about the class a implementation. Ignore the Bs. I might have confused the product tested as the whole more, I still don't jmow for certain.

I do agree with the more being class d. I think it would state if it's a/b or a/d. I was wondering just how that "pro audio" dude was telling me the ncore was a/b.
The graphs look pretty damn flat and the min/max specs look pretty damn good alsi in the documentation.

One dude tells me it's only going to sound good at low power , now someone is telling me it sounds bad at low power. Who is correct?

No, THIS isn't me.
I obviously know about the Nyquist frequency, just wanted to know if you guys were on my level. Only a stupid faggot that should kill themselves wouldn't know about it!

Good, then you'll probably know that other industries use digitally generated signals that are orders of magnitudes higher than the 44khz needed for audio, where analogue equipment would be useless.

>Sounds like something most manufacturers really wouldn't bother with.
Most don't care too much about controlled directivity either.

The obvious problem with not filtering properly can be demonstrated with MTM-style speakers (midwoofer-tweeter-midwoofer). The surround of the large midwoofers, as well as the tweeter, limits how close you can place the drivers together. Path length differences will result in strong off-axis interference if the cutoff frequency is not made sufficiently low.

>Also as far as I understand, I wouldn't consider crossovers as a compromise for poor driver pairing as you could never match drivers as effectively as you can with proper crossover design, especially active ones.
What is there to a driver pairing in the first place? Impedance and sensitivity are top priorities for a passive design, but less so for an active or digital x-over.
Frequency wiggles can be handled by standard peak EQ. Strong break up modes in stiff midrange and woofer cones would be handled either by a notch filter or a very steep cutoff filter.
Driver pairing mostly comes down to matched directivity and limiting excursion on smaller drivers. You can control most other things through judicious use of filters.

>class A and class D
That's a Devialet implementation rather than the nCore module by itself.
>now someone is telling me it sounds bad at low power. Who is correct?
Who is that someone, me? I said Class D would be more inefficient at low power, not that it would be degraded in quality. The business said so far about the topology comparison isn't much true.