"Upgrade Your Audio To 32-bit": Samsung's Audiophoolery - Page 1
Forum section: electronics / projects

author: Count Belmez
author_rank: Newbie
timestamp: December 29, 2016, 07:07:44 am
content: This is my first ever post to this forum of EEVblog. It's really my favourite electronics video blog ever! It had awesome insights about what I like to consider as "audiophoolery".Now during my boring browsing session comes something that is just utter stupidity from a big company: Samsung's proprietary tech that "upscales your audio files to 32-bit." I completely laughed as my first reaction.https://news.samsung.com/global/samsung-set-to-unveil-the-future-of-audio-at-2017-cesSo, what do you think about this?

author: DimitriP
author_rank: Super Contributor
timestamp: December 29, 2016, 07:26:46 am
content: QuoteIt was about time someone reversed engineered Mary Poppins' bag and built a sound processor out of it.I think nothing else until I can hear it playing the intro to "Hotel California".

author: DimitriP
author_rank: Super Contributor
timestamp: December 29, 2016, 07:30:00 am
content: Mary Poppins' 8 bit bag:

author: hans
author_rank: Super Contributor
timestamp: December 29, 2016, 09:21:10 am
content: If the hearing threshold is 0dB, and the acceptable threshold is 100-110dB for very brief moments.. who needs anything more than 110dB SNR?I built a USB soundcard a while ago using I2S DACs that I found using 'sort by price' on Farnell. They work fine for general use. I really can't see any point of going higher spec.

author: Fungus
author_rank: Super Contributor
timestamp: December 29, 2016, 09:24:05 am
content: Quote from: Count Belmez on December 29, 2016, 07:07:44 am"Overall, the MS750 Soundbar provides a deeper sense of immersion. This is a result of superior up-firing capabilities and overhead sound that uses vertical tweeters with upmixing technology."Translation (I think): 'Our tweeters point upwards'.Quote from: Count Belmez on December 29, 2016, 07:07:44 am"Samsung’s proprietary technology that enables 32bits of rich, detailed sound from any source that provides 8 to 24bit sound."We can all go back to 8bit sound? That's good news for people with limited bandwidth.

author: Fungus
author_rank: Super Contributor
timestamp: December 29, 2016, 09:28:25 am
content: Quote from: hans on December 29, 2016, 09:21:10 amAnybody who mixes or processes sound, obviously. Samsung's new Distortion Cancelling technology reduces sound alterations by predicting the movement of internal speaker units in advance. They'll certainly need 32 bits for that. Who are we to doubt them?

author: FreddyVictor
author_rank: Regular Contributor
timestamp: December 29, 2016, 09:34:32 am
content: Isn'tDSD1bit ?so by Samsungs theory, must be terribleusual marketing hyperbole will no doubt persuade some to depart with their hard-earned cash ...

author: wraper
author_rank: Unknown
timestamp: December 29, 2016, 10:00:03 am
content: Quote from: Fungus on December 29, 2016, 09:24:05 amI'll translate that for you. It accepts 8-24 bit data and has 32 bit DAC.

author: BrianHG
author_rank: Super Contributor
timestamp: December 29, 2016, 10:38:02 am
content: Quote from: wraper on December 29, 2016, 10:00:03 amSamsung doesn't have a 32 bit DAC, it's only 1 bit PWM like most, with 32 bit processing & dithering. Hey, but didn't they say their speakers were wireless, meaning compressed audio. Isn't compressing audio, most likely decompressed from another compressed format, then, re-compressed for their wireless format 'Anti-Audiophile'? After all, 32bit stereo, 192khz full audiophile quality, + some overhead, error correction & transmition retries, caching, eating up 25 megabit eat up valuable wireless bandwidth, I mean if it were Audiophile uncompressed full quality audio networked.Ok, maybe they didn't go this far, but the whole notion of Audiophile should be to represent the source sound to it truest actual nature without processing, engineering a speaker which is physically designed to beautifully cover the audio spectrum without corrective audio processing tricks, having a minimal single-ended class A audio amp with no ICs, just the minimal few mosfets from input to output, with a power-supply at something like 1.5x-2x the power required of your output demands, with the source 1bit pwm, or preferably a multibit DAC at 1vrms always running at the full original data volume with it's output divided down by a passive resistor volume control giving that infinite analog resolution at any volume?Today, no major consumer retail electronics company has any claim on being Audiophile, while, sadly, there are all those who claim and would seem to be audiophile who charge ridiculous $$$ for BS and lies. Some of which don't even hold true to the above paragraph which only dirties the Audiophile term even more...

author: RogerRowland
author_rank: Regular Contributor
timestamp: December 29, 2016, 11:10:24 am
content: Whether it's BS, bollocks or whatever, it doesn't actually matter to me. I only have 4-bit ears.

author: chris_leyson
author_rank: Super Contributor
timestamp: December 29, 2016, 11:16:01 am
content: Looks like Samsung are trying to market snakeoil.QuoteThe dynamic range of human hearing is often stated as being 120dB, that would be the absolute dynamic range between the threshold of hearing and the threshold of pain. The threshold of discomfort is probably a lot less than 120dB, maybe 100dB or less and that would depend on the listener. They seem to have forgoten about real world psychacoustics and auditory masking effects. The human ear is NOT a distortion free linear transducer with 120dB ofuseabledynamic range. Under auditory masking conditions the real dynamic range of human hearing might be 20 to 40dB ? Best guess. 32-bits, marketing wank for audiophools.

author: timb
author_rank: Super Contributor
timestamp: December 29, 2016, 11:37:43 am
content: Quote from: BrianHG on December 29, 2016, 10:38:02 amIt can be lossless and still wireless. FLAC and ALAC are both lossless codecs and will transmit just fine over an 802.11g link.In fact, Apple's AirPlay system uses ALAC to transmit lossless audio from iOS/macOS devices wirelessly. (With the AppleTV and AirPort Express being the official wireless receivers, though a Router, Raspberry Pi or other Linux based device with a free USB port and network connectivity will work just as well.)

author: wraper
author_rank: Unknown
timestamp: December 29, 2016, 12:04:19 pm
content: Quote from: BrianHG on December 29, 2016, 10:38:02 amAre you a true R-2R purist? Multibit DAC is basically dead for audio. Why delta-sigma DAC cannot be called 32 bit is beyond me. It has that resolution, end of the story (real ENOB is another thing of course). After all, you don't see 1-bit PWM in MCU datasheets.EDIT: BTW 1-bit DAC is just a marketing term for consumer electronics.

author: SeanB
author_rank: Super Contributor
timestamp: December 29, 2016, 12:36:48 pm
content: 32 bit, because 16 bit left channel, plus 16 bit right channel, is 32 bits........Of course ignoring that 99% of all audio DAC's are really 14 bit ones, and most of them are not at all guaranteed to be either monotonic, linear or have no missing codes with more than 10 bits at all, and those that are really 16 bits tend to have a price tag in the 4 figure range, in quantity, and also are either laser trimmed, or have calibration needed.Then you put it through some JRC opamps, or the clones that unlce Lo made in his garage, and send it via Sanwa or Teapo caps of dubious rating, to a speaker that on a good day might come close to approaching the spec of "strangled Chipmunk" and be as linear as a ball of steel wool.

author: KhronX
author_rank: Frequent Contributor
timestamp: December 29, 2016, 03:01:56 pm
content: ... And then there's the bare-walled flutter-echo-ey living-rooms these *ahem* audio-systems go intoBecause sure, soundbars are the be-all-end-all of hi-fi audio. That's why PA's "only" use 8-12" multi-speaker line-arrays and van-fulls of 18-21" subs. Dumbasses, what do they know

author: ConKbot
author_rank: Super Contributor
timestamp: December 29, 2016, 03:03:50 pm
content: Forget the DAC, I've seen the analog gubbins for a 14.5 bit ENOB data acquisition system that was good for 200kHz. And it's not small. 1 channel would take about half of a sheet of letter/A4 paper. Though selectable filters probably did blow the space up a good bit.

author: xani
author_rank: Frequent Contributor
timestamp: December 30, 2016, 07:30:17 pm
content: I wonder if there is any producer that took the piss on audiophiles and edited the "24 bit, 96kHz audiophile" quality to contain some easter eggLike use 8 LSBs to encode 8 bit audio of someone repeating "audiophiles are idiots" in loop over whole track

author: Circlotron
author_rank: Super Contributor
timestamp: December 31, 2016, 10:43:26 pm
content: "With over ten consecutive years of market leadership in home audio"Um.. no.Seen plenty of Samsung TV sets in the junk mail catalogues but never any audio stuff. Ever.

author: SL4P
author_rank: Super Contributor
timestamp: January 01, 2017, 02:27:25 am
content: QuoteIn the early digital video world, we called this concatenation (of processes)... to be avoided at almost any cost.The reason is simple. Aliasing between bit-rates, codecs, data length, frame sizes and every other variable simply leads to the introduction of unrecoverable artefacts.Even starting high, then downsampling has its drawbacks... butnevergo the other way!

author: eeFearless
author_rank: Contributor
timestamp: January 02, 2017, 06:09:14 am
content: An Audio Engineering Society paper by Meyer and Moran showed that humans (most all) can not perceive quality beyond the bit depth and sampling rate of CD quality. You need to be a member to access the paper. But, you can get a summary from a Mix magazine article online. Google "the emporer's new samplify rate".

author: NiHaoMike
author_rank: Super Contributor
timestamp: January 02, 2017, 06:48:29 am
content: Quote from: hans on December 29, 2016, 09:21:10 amUsable SNR is often lower than spec SNR. For example, if you have a headphone DAC/amp with a 100dB SNR spec and have the volume set to something like 40dB below full, the usable SNR could be as low as 60dB if the main source of noise is after the PGA.Something similar happens with most digital amplifiers where lowering the volume setting causes a loss of effective resolution. A few vary the output voltage rail with the volume setting to mitigate that and reduce power dissipation.

author: eeFearless
author_rank: Contributor
timestamp: January 02, 2017, 06:59:07 am
content: Usable SNR is often lower than spec SNR. For example, if you have a headphone DAC/amp with a 100dB SNR spec and have the volume set to something like 40dB below full, the usable SNR could be as low as 60dB if the main source of noise is after the PGA.Something similar happens with most digital amplifiers where lowering the volume setting causes a loss of effective resolution. A few vary the output voltage rail with the volume setting to mitigate that and reduce power dissipation.Most audio codes do not decrease the digital signal (before the DAC) when turning down the headphone volume. They attenuate the analog signal after the DAC (or, use less drive). So the effective resolution is preserved.

author: eeFearless
author_rank: Contributor
timestamp: January 02, 2017, 07:08:51 am
content: Consider the recording factors. A kickin' mic is around 82 dB SNR @ 1Pa (12dBA). Keep in mind that a jackhammer @ 1m is about 2Pa. So, for most practical purposes, simply the self noise of the mic will limit the dynamic range long before the ideal DR of the ADC bit depth (beyond 16 bits).

author: NiHaoMike
author_rank: Super Contributor
timestamp: January 02, 2017, 04:57:49 pm
content: Quote from: eeFearless on January 02, 2017, 06:59:07 amThe PGA is what adjusts the volume. If the main source of noise is after the PGA (e.g. the output opamps), turning down the volume decreases the signal but does not decrease the noise, lowering the SNR.

author: slicendice
author_rank: Frequent Contributor
timestamp: January 02, 2017, 08:31:41 pm
content: There's a big difference in having a 32-bit ADC vs a 32-bit DAC. In ADC (input) it is very useful, but for DAC (output) it's totally useless, even 24-bit is overkill. There are a lot better techniques to get good SNR for playback.HEREis a good article on this which explains the differences.

author: 0b01010011
author_rank: Regular Contributor
timestamp: January 04, 2017, 10:24:56 am
content: These are the same Samsung that have a valve '1.3kW*' 5.1 speaker system running off of 2 minature valves...http://www.samsung.com/uk/consumer/tv-audio-video/audio-video/home-theatre-systems/HT-J7750W/XU

author: Fungus
author_rank: Super Contributor
timestamp: January 04, 2017, 11:30:42 am
content: Is this as bad as "4K Ultra HD audio"?http://www.bestbuy.com/site/4k-ultra-hd-ecosystem/4k-ultra-hd-audio/pcmcat748301920615.c?id=pcmcat748301920615

author: slicendice
author_rank: Frequent Contributor
timestamp: January 04, 2017, 01:40:41 pm
content: What? Are you saying 4K Ultra HD Audio is a bad thing? That's just a device that splits your audio from any devices/players, connected to it, to your audio system and sends the rest (the 4k video) to the UHD TV.Google about HDMI standards and you'll understand why there is a big difference between a device that only support FHD vs a device that support UHD. There is a huge bandwidth requirement difference in the 2 standards. :-)

author: BrianHG
author_rank: Super Contributor
timestamp: January 04, 2017, 02:02:29 pm
content: Quote from: slicendice on January 02, 2017, 08:31:41 pmOk, say I have my overkill 24bit DAC driving my 1KW/channel, 4kw total audio amp with the data scaled down to -100db for normal comfort listening. How many bits in amplitude is the preamp feeding the 24bit dac for that -100db level output?

author: slicendice
author_rank: Frequent Contributor
timestamp: January 04, 2017, 02:54:35 pm
content: hahah, is this a math contest? Does it matter? Your ear could not distinguish between the high lvl and low lvl sounds lost anyway at that low level of volume. Maybe if I was a Ferret or something else that can hear a huge range (ferrets can hear sound from about 10Hz to 100KHz) of frequencies it would make a difference, but not for a human.And why on earth would I have a 1KW system in my living room in the first place? Speaking of overkill!I have a friend who actually has a 2.5KW system at home. Totally useless for his purposes. Unless he wants to make his friends ears bleed their heart to stop and their guts to explode. Maybe he secretly dream of killing uninvited guests with his AMP?Those devices are good for destroying the structural integrity of your house, meaning it's not safe to live there after playing at full volume.

author: slicendice
author_rank: Frequent Contributor
timestamp: January 04, 2017, 03:22:45 pm
content: Here's a hint:Minimizing noiseThe key to minimizing noise is to keep the digital signal level as high as possible as it passes through without overdriving the audio circuits. The speaker volume should be turned up just high enough to reach the loudest undistorted volume that you would normally listen to. Maximizing your amplification and minimizing your digital signal will end up in a lot of undesired noise in relation to the actual digital signal you are passing through.

author: Watth
author_rank: Frequent Contributor
timestamp: January 04, 2017, 03:45:37 pm
content: Wouldn't that make sense in the case of digital processing?Like in the case of a bitmap image treatment you'd better quadruple the number of points, make your adjustments, then rescale the image to its original size?

author: Richard Crowley
author_rank: Super Contributor
timestamp: January 04, 2017, 04:01:24 pm
content: I have read reports that the kind of music most people listen to these days (240~320Kbps MP3) is effectively 14 or 15 bit by the time it is decoded and reaches your ears. Even "conventional" 16-bit is a dynamic range of 96dB which is pushing it for most mass-produced consumer audio gear.We prefer to record (actually "track") live performances using 24-bit depth just to have the headroom to accommodate over-exuberant peaks that exceed the rehearsal level-setting. 24-bit equates to 144dB dynamic range which rather exceeds the 120dB noise floor limits of even most professional audio gear.32-bit equates to 192dB which I believe is quite beyond what can be produced in the average atmospheric pressure on this planet.

author: NiHaoMike
author_rank: Super Contributor
timestamp: January 04, 2017, 04:17:38 pm
content: Quote from: slicendice on January 04, 2017, 01:40:41 pmThat exact configuration is exactly what gamers tend to dislike because the receiver may add lag. Ironically, that's more likely with the higher end receivers that try to "do it all". What I would like to see is a shrunk down amplifier that's designed to mount behind the TV and gets its input via S/PDIF or HDMI ARC or a device that is similar to size to existing receivers but is also a gaming HTPC. TI has made a 600W amplifier reference design about the size of a deck of cards for many years already. There really is no need to have receivers oversized for what they do.

author: slicendice
author_rank: Frequent Contributor
timestamp: January 04, 2017, 04:24:21 pm
content: In audio devices I would not cripple the digital input signal by lowering the "volume" of the digital signal. That would decrease the amount of effective bits available for the DAC to process. In the case we would want to go that route we would lose the dynamic range no matter how many bits we have in the DAC. Even a 1024-bit signal would become crippled. Audiofools would love that 1024-bit DAC and sell their soul for such crap.A better way is to lower the voltage of the amplification output. An amplifier is a voltage follower. Which means that the analog signal coming out of the DAC will be amplified n-times.Another way to lower the volume is to lower the voltage levels of the analog signal that comes out from the DAC, but if we have turned amplification to MAX and just lower the converted signal we will end up having a lot of noise in the amplified signal that will in worst case scenario cancel out the actual sound we want to hear. The sound would disappear into the noise. So why would we even consider doing that?24-bit is just a convenient number that works well with computer systems and similar. 24-bits = 3x1byte. 1byte = 8-bit = 256 different signal levels. 24-bit = 16777216 different signal levels in audio while the same bits would be the same millions of colors in a picture. 17million is plenty of datapoints we can recreate the original analog signal from. The output of the DAC will look exactly the same as the original input to the ADC.The dynamic range of human hearing is roughly 140db depending on frequency. A 24-bit DAC could theoretically decode roughly 144db (6db/bit). So someone explain to me what we need that extra 8-bits for in 32-bit DACs? Do we need 192db dynamic range, when we can hear only to about 140db in best case scenario?I am getting older so for me that range is closer to 100 than 140.Edit: corrected a few numbers where 144 was supposed to be 140

author: NiHaoMike
author_rank: Super Contributor
timestamp: January 05, 2017, 04:47:33 am
content: Quote from: slicendice on January 04, 2017, 04:24:21 pmModern digital amplifiers don't work like that. They use Delta Sigma to directly PWM the output stage. The best designs do vary the supply voltage with the volume setting.Even for headphones, the old DAC/amp design is slowly being displaced by single chip digital amplifiers. Things have literally gotten to the point where removing a lot of the analog circuits and replacing it with a bit of digital logic nets a lower power usage. Smartphones going USB-C are the main driver behind that development.

author: slicendice
author_rank: Frequent Contributor
timestamp: January 05, 2017, 08:16:55 am
content: Yes, of course you can use analog or digital adjustment of the output signal, but that has nothing to do with the DAC being 24-bit or something else. You don't remove any bits from the digital audio signal it self in order to adjust the volume. The purpose of a DAC is to convert the digital numbers that represents voltage levels at a specific time into a smooth analog signal that is amplified so it can drive the speakers.Why are we now confusing DAC with AMP? They have 2 completely different purposes and has nothing to do with audio quality when going from digital to analog.Sure one could build ADC, DAC, AMP with PWM into one chip for efficiency, but that is beyond the scope of this thread and still has nothing to do with the digital signal being 24-bit or 32-bit.EDIT: We can talk about differences in CLASS A, B, A-B and D amplifiers in a separate thread, but please leave it out of this thread as it has nothing to do with the DAC

author: NiHaoMike
author_rank: Super Contributor
timestamp: January 05, 2017, 03:58:32 pm
content: A digital amplifier *is* a DAC. In fact, they're sometimes called "power DACs". Conceptually, it's just taking a Delta Sigma DAC and replacing the output stage with one that can handle a higher power. Those that don't have a variable supply rail (all the cheaper units) can only adjust the volume in the digital logic that generates the PWM.Try taking a Delta Sigma DAC and connecting a pair of headphones to the output using just some blocking caps and maybe some current limiting resistors. You'll probably be surprised at how decent it plays, maybe too loud even. (The signal into a pair of headphones at normal listening volume is tiny, on the order of -20dBm or less.) Going from a DAC to a "digital headphone amplifier" is then just a matter of adding a charge pump to allow it to swing negative (eliminating the bulky blocking caps) and a few other tweaks like some logic to prevent a glitch when powering up.

author: slicendice
author_rank: Frequent Contributor
timestamp: January 05, 2017, 04:18:11 pm
content: lol, so what if a digital amplifier (Power AMP) can drive speakers directly, still has nothing to do with bit depth in an ADC, which is the topic of this thread.So please, rather than trying to be right about what kind of amplifier is the right one to use, instead give your opinion on the "is 32-bit in a DAC needed?" question.

author: Fungus
author_rank: Super Contributor
timestamp: January 05, 2017, 04:55:22 pm
content: Quote from: slicendice on January 05, 2017, 04:18:11 pm16 bits is enough forplaybackto humans.The "volume control" argument is a red herring IMHO, people usually don't have their amplifiers/speakers set to produce anywhere near 96dB of dynamic range.44.1kHz sample rate is OK but quite close to the limit. 48kHz doesn't cost anything to implement these days, so... why not?96kHz, 24 bit? You could argue the case for recording/mixing/mastering butnotneeded for playback.More info:https://xiph.org/video/vid2.shtml

author: slicendice
author_rank: Frequent Contributor
timestamp: January 05, 2017, 05:27:19 pm
content: Quote from: Fungus on January 05, 2017, 04:55:22 pmyes closer to 16-bit than 32-bit. I think 24-bit is just fine as it gives a bit more wiggle room.8-bit was producing bad sound, 16-bit made it really good, and 24-bit put the cherry on the top. I don't think the average person can hear much of a difference between 16-bit and 24-bit for most music styles(with a few exceptions).An audiofool hears a difference in 24-bit and 32-bit, that's for sure, but it has nothing to do with the actual reproduced sound quality, but rather with whats happening in the audiofools head.

author: Fungus
author_rank: Super Contributor
timestamp: January 05, 2017, 05:39:04 pm
content: Quote from: slicendice on January 05, 2017, 05:27:19 pmNot really true. Compact cassettes (if you remember those) were only about 6 'bits' and they sounded OK to a lot of people. The lack of 'bits' is what caused the tape hiss.Quote from: slicendice on January 05, 2017, 05:27:19 pmWith 16 bits the 'tape hiss' is already inaudible.Watch the video I posted above, it shows exactly what lack of bit depth actually does to the sound.

author: NiHaoMike
author_rank: Super Contributor
timestamp: January 06, 2017, 02:23:12 am
content: Quote from: slicendice on January 05, 2017, 04:18:11 pmOn a digital amplifier (power DAC) with a fixed supply rail and 24 bits into the modulator, you'll only be able to turn down the volume 48dB below full scale if you want at least 16 bits of usable resolution. I wouldn't be surprised if many of those digital amplifier DSPs already run 32 bits into the modulator, since the DSP core itself is likely 32 bit. Actually running 32 bits into the DSP is probably of not much use.Higher sample rates definitely have their uses. Not so much for the user to hear but rather for devices to hear. As in software based sonar for gaming and whatnot. Or superimposing ultrasonic tones onto a speakerphone VoIP conversation and then using that to tune the anti echo filter in real time. Or as a low bandwidth, short range data channel for setting up encrypted wireless links. (Chromecast does that for guest mode connections.)

author: pitagoras
author_rank: Regular Contributor
timestamp: January 06, 2017, 02:31:57 am
content: I love when non-engineering-sound affirmations like samsung's are made over engineering-sound affirmations (like huge digital audio theory). Forget it, it's marketing.

author: slicendice
author_rank: Frequent Contributor
timestamp: January 06, 2017, 09:15:59 am
content: Quote from: NiHaoMike on January 06, 2017, 02:23:12 amWhat on earth are you trying to explain here? Sample rate has nothing to do with the bit depth of a DAC. You need a faster DAC for that not more bits. bit depth is all about dynamic range and unwanted noise.Watch the video Fungus posted, and you see why high bit depth can be useful when eliminating noise.A sample rate is how often you have a control points in your data. The more samples the the less data you lose in cases where the waveform has fast rise time, thus improving quality.At 8-bit one can reproduce the original analog signal from digital, but there will be more noise than using 16-bit. It is possible to get rid of most of that noise, but it's almost not audible.The data is moved digitally and without any loss (ones and zeroes) until it's time to hear the data, then it's converted to an analog signal. Whether it's amplified to drive speakers within a chip or externally does not matter, the end result is the same.

author: KhronX
author_rank: Frequent Contributor
timestamp: January 12, 2017, 12:57:58 pm
content: Weeeell...https://youtu.be/BYTlN6wjcvQ?t=2748It should start at 45:48 in.

author: slicendice
author_rank: Frequent Contributor
timestamp: January 12, 2017, 02:10:02 pm
content: Quote from: KhronX on January 12, 2017, 12:57:58 pmAwesome video snippet, that reveals all you need to know about what happens when the bit depth is reduced.

author: KhronX
author_rank: Frequent Contributor
timestamp: January 12, 2017, 02:12:55 pm
content: When i read the stuff re: bit depth here, that's the section i instantly got reminded ofTook a few minutes to find it, in that hour-long video though

author: JanJansen
author_rank: Frequent Contributor
timestamp: January 12, 2017, 04:28:07 pm
content: Quote from: eeFearless on January 02, 2017, 06:09:14 amThat is not true, you hear big difference between CD quality and 96k 24bit.

author: Richard Crowley
author_rank: Super Contributor
timestamp: January 12, 2017, 04:51:31 pm
content: Quote from: JanJansen on January 12, 2017, 04:28:07 pmIs that just a deeply-held religious belief, or can you cite authoritative research?

author: CatalinaWOW
author_rank: Super Contributor
timestamp: January 12, 2017, 05:12:13 pm
content: Quote from: JanJansen on January 12, 2017, 04:28:07 pmMaybe you should rephrase that to say that some humans on some carefully selected recordings can tell the difference. I know I can't. On any piece of music that I have tried. The scholarly paper admitted that some people can.Audio is weird stuff. I transcribe old records (mostly LPs) to digital media and use Audacity to clean up noises from scratches, dirt and the like. Tedious stuff, but worth it on music I like. Watching the waveforms I will see huge spikes and discontinuities that I can't hear on the reproduced sound. And then spend tens of minutes finding some tiny wiggle that makes an offensive sound in the output. I haven't been able to find an algorithmic description that matches these observations, so haven't been able to automate the process.

author: Fungus
author_rank: Super Contributor
timestamp: January 12, 2017, 05:20:16 pm
content: Quote from: CatalinaWOW on January 12, 2017, 05:12:13 pmWhich humans, exactly? Which recordings? Citation needed.Nothing about the human ear suggests it might be possible to do that.

author: Buriedcode
author_rank: Super Contributor
timestamp: January 12, 2017, 05:23:00 pm
content: To muddy the waters often higher bitrate releases are 'remastered' versions of pieces/songs. This makes fair subjective comparison pretty much impossible, since they are not playing the same source, and who bothers to play back 24-bit/96kHz audio at 16-bit 48kHz?. I cannot claim to hear a difference between 24-bit/96kHz and CD quality, but in certain circumstances I can hear when something has been re-sampled - but this could well be just a poor re-sampling.I am regularly amazed at the sort of stuff that's claimed to be objectively true, with little evidence for it. Its not just that people want to believe something, they genuinely believe its true and that there's a wealth of evidence for it, even though there isn't. Because the idea its somehow wrong is so offensive, it is assumed that all evidence points their way. All-in-all the reason there's so much marketing bollocks is because it can be extremely difficult to verify strongly held beliefs. If one runs a very strict analysis on say, CD quality, and higher bit-depth, if they dont' find any real difference, then of course one could say "yes but I *hear* a difference!" which cannot easily be refuted because we will never know exactly *what* someone else hears - that and the fact 'reality' is just a combination of input from multiple senses and what's *expected* by the mind. That's why taste is influenced by sound, and sound influenced by sight, people gauge the weight of an object differently if they see someone struggling to pick it up before they try etc..With that ambiguity the claims can't always be refuted and so.. take hold.. which re-enforce belief's, which then feeds back in to demand, and therefore, more marketing for that particular gimmick. The audio and cosmetic industries are prime examples (makes you *look* younger.. with the only metric being someones opinion).I am not suggesting that someone cannot tell the difference between the same source at 24-bit/96kHz and 16-bit at 44.1kHz (I can't) just that it is very easy to be influenced as to what one hears.And just before I step off my soap box: something I regularly say to those I know with thousands of pounds of 'high quality' audio gear - if the quest for perfection, or being 'annoyed' at hearing 16-bit 44.1kHz audio in music is what drives you - you clearly don't enjoy music. Surely it is more important to listen to music, rather than dwell on what formats you think you should listen to. Give me MP3 over FLAC any day, chances are I'll be listening with ear buds on the way to work, so I won't notice much of a difference, and I'd prefer to have access to lots of different music, than a selection of 'super high quality' ones. A good song is a good song even if its played on a radio.

author: Fungus
author_rank: Super Contributor
timestamp: January 12, 2017, 05:32:47 pm
content: nb. CD mastering can definitely be messed up. I once managed to scratch one of my favorite CDs and when I bought another one the CD recording was terrible. Clipping, distortion, you name it. The original one was much better sound. Whoever 'mastered' the second batch needs firing, it was unlistenable.I went on eBay to see if I could find anything and found an "MFSL" remastered SACD. MFSL labs supposed take old recordings and pass them though a huge machine to remaster them in 96k. The SACD disk has both normal CD and SACD in a layer underneath. It's a prestigious lab, should be good, right?Wrong! The CD version had no treble at all. They'd deliberately removed it, presumably so they could switch between the two and say "Listen to the massive difference 96k makes!" (see image)How did I solve it in the end? With a 44.1kHz FLAC file downloaded from The Pirate Bay. Problem solved!

author: Siwastaja
author_rank: Super Contributor
timestamp: January 12, 2017, 05:35:18 pm
content: Oh, these kind of topics are kind of a litmus test on general audio clue(less)ness. When anybody claims there is any audible benefit on music distribution at 24 bits (like "puts a cherry on top of 16 bits"), or that they can hear a clear difference between the CD specs and something over it, you can instantly see they are utterly and totally clueless about audio and sound fidelity, and you can easily ignore them in any audio-related questions in later discussions.They are not necessarily lying nor stupid in general; audio, especially digital audio, just isn't within their field of understandingat all. When they hear differences, there either isn't a difference to begin with, or when there is, it's caused by something totally different to what they are thinking about (typically the original source being totally different!), but they are not "cluefull" or understanding enough to spot where their assumptions are going wrong, because it's not always trivial.

author: Fungus
author_rank: Super Contributor
timestamp: January 12, 2017, 05:38:00 pm
content: Quote from: Buriedcode on January 12, 2017, 05:23:00 pmI could tell tales of people sending CDs off to be duplicated and the first thing the 'engineer' does is transfer the CD to a DAT via an analog RCA cable. Why? Because that's what the CD pressing machine uses!(that was back in the 1990s)

author: Fungus
author_rank: Super Contributor
timestamp: January 12, 2017, 05:40:43 pm
content: You have to be very careful when comparing audio. A tiny difference in volume level makes a huge difference in perceived sound.

author: Fungus
author_rank: Super Contributor
timestamp: January 12, 2017, 05:58:54 pm
content: Quote from: Buriedcode on January 12, 2017, 05:23:00 pmMy answer to you is that in that case there's no point in going to live concerts, you can hear exactly the same song at home.I don't necessarily mean loud rock+roll concerts either. Anybody who's listened to a busker in the street or classical music in a good concert hall can tell you it's a world apart from listening at home. There's a reason they put all those wood panels in the ceilings of opera houses, etc.b) It also wouldn't matter what equipment you use to record music, you might as well just use the cheapest microphone possible, right? Nope, even the most "you clearly don't enjoy music" proponent will see the sense in using good recording gear, in putting egg boxes on the walls to avoid echos, etc.Your argument is

author: slicendice
author_rank: Frequent Contributor
timestamp: January 12, 2017, 06:11:46 pm
content: For the audio demonstration where the bitrate was dropped from 24 down to 4, I could hear the background hiss at around 19-bits, but at 16-bits the music sounded still good, though a lot more hiss, at around 12-bits I heard some distortion/clipping or whatever one could call it. Though I could clearly hear the difference, I am certain 32-bit compared to 24-bit will not make any difference at all for playback, no matter what recording it is.And yes, I DO listen to classical music, and NO a home AMP and speakers can NEVER reproduce the sound at a real concert. Does not matter how much the speakers and amps cost, or how many bits the DAC has, not even analog amps can reproduce it.BUT that is completely different scenario. We can't compare a concert hall with 16-bit or 32-bit DACs. Maybe with a 1024 speaker surround system we can start talking about concert hall accuracy. For such accuracy we need to reproduce the environment, not get better speakers or higher lvl bit depth in the DAC.

author: Fungus
author_rank: Super Contributor
timestamp: January 12, 2017, 06:28:51 pm
content: Quote from: slicendice on January 12, 2017, 06:11:46 pmTrue, but trying to move a tiny bit closer towards that sound isn't pointless though.Have you ever been to a hi-fi show? Had a demo in a decent Hi-Fi shop? Thereisa difference, there's no question about that.The trick is to distill out the phoolery and buy sensible gear. These days you can get decent electronics for $1000. That leaves the speakers, you can can get some very good speakers for under $2000. After that you're probably better off spending money on sound conditioning your listening room, etc. This lets you actually hear what you've paid for.

author: Buriedcode
author_rank: Super Contributor
timestamp: January 12, 2017, 06:32:02 pm
content: Quote from: Fungus on January 12, 2017, 05:58:54 pmle sigh. A live song will always be different from the Album recording (unless they mime). Plus, people dont' go to gigs to hear "exactly the same song", they go to *see* the band play. And a live busker playing a song again, with be different from any album recording - that is what sets it apart. I'm not sure why you are trying to compare the two?And the mention of recording gear.. I was referring to playing the same album from MP3 or FLAC, with speakers, or headphones. Of course a terrible recording can be detrimental to the enjoyment.. but I was talking about playback, because that is what this thread is about. Of course recording is done with the best possible quality, because you can't lower the noise floor or improve quality with post processing, only lose quality or increase noise, so they want a perfectly quiet, dry recording with headroom to allow for endless processing. This is why the push for 24-bit 96, even 192kHz is used in the industry. Your point is mute on the subject of playback quality.My point was.. again.. listening to and enjoying music is surely more important than restricting yourself to only listening to music on specific gear, or at a specific bit depth.

author: Buriedcode
author_rank: Super Contributor
timestamp: January 12, 2017, 06:36:51 pm
content: Quote from: Fungus on January 12, 2017, 06:28:51 pmSo, if one just plays music, from FLAC a CD or MP3 on their phone, or PC, they aren't really hearing the album they paid for? Although I know it isn't the case, but it almost seems like you will only ever listen to music in a room dedicated to acoustics, with matched speakers, and specific equipment. If that's your thing, good for you, but I tend to listen to music a lot.. and frankly, I settle for my phone or £15 headphones because I'm too busy playing along to it..

author: slicendice
author_rank: Frequent Contributor
timestamp: January 12, 2017, 07:21:11 pm
content: Quote from: Buriedcode on January 12, 2017, 06:36:51 pmI think you misunderstood what he meant...? That is not what he was saying. After you have found the "optimal" gear that can reproduce what is on the recording you want to play, the next step is to make the environment suitable for such high quality playback. Another option would be a set of good headsets that can reproduce the dynamic range.There's no point in spending thousands on audio gear if your environment isn't right. Good headphones solves the audio/hearing part but there's more to music than just what you hear. What you feel when the sound waves hits you is also important. This is why you might need to spend some money on both good playback equipment and environment.For music on the go, for me any set of decent headphones is good enough and the source does not even have to be that good quality.This still has nothing to do with the fact that we really don't need a 32-bit DAC. 24-bit is more than enough for both recording and playback. Only for some rare cases we might need 32-bit or even 64-bit processing power when editing music and want to add some complex effects and adjust sound levels. But most likely that high level of bit-depth will add more noise than actually help making better sound.Here'ssomething to read about bit depth.

author: Fungus
author_rank: Super Contributor
timestamp: January 12, 2017, 07:24:02 pm
content: Quote from: Buriedcode on January 12, 2017, 06:32:02 pmYou contradict yourself. If the reproduction equipment isn't critically important then neither is the recording equipment.It doesn't need to be "terrible", just good enough. In the old days the band would just stand in front of a microphone and play. Why change that if the only thing that counts is playing good songs?Quote from: Buriedcode on January 12, 2017, 06:36:51 pmThese days I mostly listen to music on $150 headphones using a $12 USB DAC.

author: Fungus
author_rank: Super Contributor
timestamp: January 12, 2017, 07:25:36 pm
content: Quote from: slicendice on January 12, 2017, 07:21:11 pmAll it really needs is enough volume compression to remove all the dynamics.

author: Fungus
author_rank: Super Contributor
timestamp: January 12, 2017, 07:31:59 pm
content: Quote from: slicendice on January 12, 2017, 07:21:11 pmOr just go straight to the source, ie. the video linked from that article.It has some nice test gear in it, unlike the text article:https://www.youtube.com/watch?v=cIQ9IXSUzuM&feature=youtu.be

author: slicendice
author_rank: Frequent Contributor
timestamp: January 12, 2017, 07:34:51 pm
content: Quote from: Fungus on January 12, 2017, 07:25:36 pm

author: slicendice
author_rank: Frequent Contributor
timestamp: January 12, 2017, 07:41:25 pm
content: Quote from: Fungus on January 12, 2017, 07:31:59 pmJust realized this vid is from Image-Line. I LOVE FruityLoops Studio, awesome piece of software!

author: Buriedcode
author_rank: Super Contributor
timestamp: January 12, 2017, 09:19:17 pm
content: Quote from: Fungus on January 12, 2017, 07:24:02 pmI do not see the contradiction here, you're assuming that recording and reproduction is a symmetrical process - it really isn't, it is a one way process where the extra dynamic range can help prevent clipping (which of course cannot be removed once recorded) and gives 'headroom' for the degradation of signal during processing.When recording, producers and sound engineers want the cleanest, dry recording at the best possible quality, because processing inevitably introduces noise, and you can't remove room reverb or artifacts. The quality of recording - straight form the mic has to be the best, and the bit depth gives headroom for this and there is always degradation during the process so sound engineers want to start off with the best possible recording with the highest dynamic range they can get. These only benefit the recording side, and are not really noticed in playback, except that it allows for much more processing before artifacts are noticeable.Reproduction does not undergo any significant processing. Sure some EQ, but for the most part it isn't going to be subjected to layers and layers of processing that a lot of music these days undergoes is it? So for delivery, 16-bit 48kHz will suffice as there is nothing to really degrade it. I think I'm starting to repeat myself, plus, slightly off topic, and so, shall leave the thread..

author: helius
author_rank: Super Contributor
timestamp: January 12, 2017, 10:03:29 pm
content: Quote from: slicendice on January 12, 2017, 06:11:46 pmAnd yet, that sound is simply two longitudinal waves which are picked up by your ears. And you only have two ears. So why would you assume that the original sound can never be reproduced, from a two-channel recording? Isn't it more likely that instead of a physical impossibility, what you have experienced is a technical limitation?A properly recorded and reproduced two-channel recording can preserve the direction of sounds in both axes, inclination and azimuth, to the sensitivity of the auditory tract:1 degree of angle.(And, yes, this does require a time resolution higher than 44.1 kHz. From the same article: "Humans can discern interaural time differences of 10 microseconds or less.")

author: Fungus
author_rank: Super Contributor
timestamp: January 12, 2017, 10:09:51 pm
content: Quote from: Buriedcode on January 12, 2017, 09:19:17 pmNo arguments there. I thought we were arguing about the reproduction hardware.Quote from: Buriedcode on January 12, 2017, 09:19:17 pmHow is it off topic in this thread?

author: slicendice
author_rank: Frequent Contributor
timestamp: January 12, 2017, 10:28:57 pm
content: Quote from: helius on January 12, 2017, 10:03:29 pmMaybe I said it a bit wrong, but you are on the right track. You can't reproduce a 3D environmental sound completely accurately with just a stereo recording and stereo speaker system. You can mimic it but the reproduction is not 100% accurate once it hits the human ear.Anyways, this was way out of topic now. I still think that in recording/editing higher bitdepth can in some cases help a lot, but for reproducing the digital sound using more than 24bit is a total waste of decoding resources and does not improve the sound quality the slightest. And I think I have said this too many times already in this thread. Time to close the lid on this one and move on. :-D

author: Don Hills
author_rank: Regular Contributor
timestamp: January 13, 2017, 01:09:42 am
content: Quote from: helius on January 12, 2017, 10:03:29 pmIncorrect. 16/44.1 is sufficient.The time resolution of a 16 bit, 44.1khz PCM channel is not limited to the 22.7µs time difference between samples. The actual minimum time resolution is equivalent to 1/(2pi * quantization levels * sample rate). For 16/44.1, that is 1/(2pi * 65536 * 44100), which is about 55 picoseconds. To put that in perspective, light travels less than an inch in that time.Shannon and Nyquist stated that as long as you keep all components of the input signal below half the sampling frequency, you can reconstruct the original signal perfectly - not just in terms of amplitude, but in terms of temporal relationships too. They only addressed sampling, and assumed infinite resolution in amplitude. With a digital signal the precision is limited by the number of amplitude steps, leading to the above formula.

author: slicendice
author_rank: Frequent Contributor
timestamp: January 13, 2017, 07:54:35 am
content: I can't hear any difference in 44k, 48k and 96k sampling rate playback, but if the audio is recorded with higher sampling rate and I add complex filters and effects to them the difference is HUGE! But it has nothing to do with the playback, 44k, 48k and 96k sound still the same in either case at least in my ears. 22k again sound very different.

author: Fungus
author_rank: Super Contributor
timestamp: January 13, 2017, 02:21:10 pm
content: Quote from: slicendice on January 12, 2017, 10:28:57 pmUsing more than 16 is asmallerwaste, but still a waste.Quote from: Don Hills on January 13, 2017, 01:09:42 amYep. Monty's video has a very clear demonstration of reproducing a rising edge anywhere between two sample points (at the 21 minute mark).If anybody here hasn't watched it a dozen times before posting, please do so.Quote from: slicendice on January 13, 2017, 07:54:35 amI'd lean slightly towards 48k for playback.The reason is that a low pass filter is often applied at the mastering stage to bandwidth-limit the sound to the Nyquist frequency (this helps to get perfect reconstruction). A slightly higher sample rate lets you use a less aggressive filter.

author: JanJansen
author_rank: Frequent Contributor
timestamp: January 13, 2017, 02:21:37 pm
content: Ok, its true most speakers has roll off frequency to 20K analog, you cannot hear much difference, only the hihats will be less digital sounding, more clean open sounding, especially high volumes, it sounds more natural like the original analog source.Imagine a 20K sinewave on a 44K timeframe, that is horrible.Also young people can hear higher frequencys then old persons.Indeed : with processing CD quality is horrible.

author: Fungus
author_rank: Super Contributor
timestamp: January 13, 2017, 04:13:07 pm
content: Quote from: JanJansen on January 13, 2017, 02:21:37 pmNo, it can be reproduced perfectly.Quote from: JanJansen on January 13, 2017, 02:21:37 pmAlso not universally true.Someyoung people can hear higher frequencies thansomeold people.(In fact today's earbud-wearing generation are more likely to sufferhearing loss).

author: Delta
author_rank: Super Contributor
timestamp: January 13, 2017, 04:49:03 pm
content: Quote from: JanJansen on January 13, 2017, 02:21:37 pmhttps://people.xiph.org/~xiphmont/demo/neil-young.html

author: slicendice
author_rank: Frequent Contributor
timestamp: January 13, 2017, 07:11:21 pm
content: Quote from: Delta on January 13, 2017, 04:49:03 pmThis article contains everything and even more than we have been discussing here. Awesome stuff!So next time someone says we need 32-bit DAC and 192K Sampling rate for playback I will just put THIS -->emoji in my reply.

author: Fungus
author_rank: Super Contributor
timestamp: January 13, 2017, 07:29:26 pm
content: Quote from: slicendice on January 13, 2017, 07:11:21 pmSo does his video,andit has analog 'scopes in it showing that the mathreally works, what bit depth actually means to a signal, etc.Is nobody watching that?

author: Siwastaja
author_rank: Super Contributor
timestamp: January 13, 2017, 07:39:15 pm
content: Fully watching (with brain turned on) that video, posted a zillion times now, should be a prerequisite for posting to this thread.

author: slicendice
author_rank: Frequent Contributor
timestamp: January 13, 2017, 08:04:50 pm
content: Yes, yes, seen it gazillion times already! Awesome stuff!

author: JanJansen
author_rank: Frequent Contributor
timestamp: January 20, 2017, 03:37:19 pm
content: I did not wanto ruin your technical topic,so here is the new topic :https://www.eevblog.com/forum/projects/digital-audio-quality-discussion/please continue, say it : samsung is no good!, cartel prices and all, bribes, etc.