@@cjsvega Do not use an Intel Celeron CPU, they are junk mainly for office use, like documents, spreadsheets, etc, you need a Pentium D or Pentium 4, Dual Core ideally a 2.0GHZ CPU or higher, or an i7 quadcore CPU such as i7 920 CPU.
@@speedyboishan87 no you don't, a music server playing FLAC ,wav or even mp3 will run just fine on just about any PC ever made to run Windows 95. and that's prehistoric technology, but would still work. winamp was free music server software thats still beats newer aps.
What a pleasure to watch informative videos presented by someone mature who's not trying to impress or be cool. Just straight down the line engineering. You are the duck's nuts of TH-cam Paul 👏
sädle he didnt know if his chippäir amp wöz 200? ör 2k. well was ja maybe it was 2k 4k 42 ? but almost 5050 it was an extra diggit v$v 1:15 nö aströhöbbyy well nö ge xD
I intensely stare at the cd for about an hour and then write 1s and 0s in notepad++, save, and rename to wav. (Edit:2020) During the lockdown i have mastered the art of engraving CD by hand. I have now achieved replica quality of the original content. If the lockdowns get extended i might go so far to manually magnetize hard drive platters in order to achieve faultless source migration. P. S. You people are one helluva community
@@arthurwatts1680 WAV files do not include checksums (beyond that provided transparently by the media hardware such as disk sector CRCs), but when they are encoded for transfer to an audio CD, error detection/correction data is added.
@@marianneoelund2940 Thanks for the clarification, Marianne, but I think you'll find that Mr Jerkovic (!) and myself were just trying to inject a little humor into what is a very dry topic. There was a time when I agonized over what EAC was doing with my CD rips but it's been 6 years since I've purchased a CD so its all a bit beyond me now. I know - flat-earthers and all that - but it is what it is. MQA and DSD are equally irrelevant in my little world, fwiw, but I realise that the world doesnt revolve around moi.
@@arthurwatts1680 I appreciate that. But I often take opportunities like this to post trivia which might be interesting to someone. The lack of checksums or CRCs in most audio files makes them rather easy to hack or modify at the bit level - something I've been taking advantage of recently.
As a computer scientist i can say flac and wav are identical... if you are worried about the size reduction, dont be. As for the possibility of sounding worse due to the extra processing to uncompress, i see your reasonining, but i see that as an extremely rare case! If that happened to you, your dac has a really really major flaw, because decompressing the flac is super simple stuff, just throw away that dac!
What everyone here seems to be missing is the fact that even Paul said the decoded bits are identical. In other words, he agrees with most of these arguments in the comments. I believe the confusion comes from the fact that he is often digressing to address the entire signal path from the media to the speaker output rather than the differences in the file format. Even though the decoded bits are identical, it is easily possible for the final analog waveform to vary noticeably. The differences people can hear are not due to changes in the bits. Like everyone (including Paul) keeps saying, the bits are identical. The difference in sound comes from other factors. First, there can be minor changes in the amount of time between the bits during the playback. (Not decoding.) These tiny differences actually DO alter the final analog output signal which is created from the bit stream. An extremely accurate waveform comparison between the analog output signals being sent to the amplifier would show those differences. When the width of the pulse changes, the waveform changes. That's the entire premise of Pulse Width Modulation. Another contributing factor is the hardware and software involved in processing the identical bits. Most audio circuitry does some level of filtration at various stages during the conversion process back to analog. These filters assume that the amount of time between the bits conforms to the sample rate perfectly, which would (in theory) produce an exact replica of the original waveform. This is often not the case. Digital filtration, in particular, is highly timing dependent. If the spacing changes, the effect of the filter will also change. Analog filtration can also change the sound since it is based on the controlled attenuation effect of capacitors and inductors responding to varying frequencies and amplitude within the signal. Then, there is the fact that the actual values of electronic components vary randomly within a specific tolerance range, which will also affect the output. I'm quite certain that there are myriad other factors along the signal path which can slightly alter the analog output in a way that affects the fidelity of the output waveform to the originally recorded waveform. So, the bottom line is that, although the bits are precisely identical, the output can indeed be physically (and even audibly) different due to the influence of other factors. What really matters is whether you enjoy what you hear, and that is purely subjective. So, why pick on others who love music as much as we do?
Thank you, but having said all this, surely there is a process which would ensure all these errant bits remain in or can be returned to their correct positions ?
I Rockbox'd my old mp3 players, in order for it to play FLAC & other lossless format. The weak processors in these device most certainly decode FLAC in real time, even huge merged audiobook files. The open source firmware let's user peek the computation as the file plays. One has to be running something ancient to not practically decode level 8 FLAC
44.1Khz was selected because originally the audio was processed in one unit and storage was a video tape recorder. Originally 3/4" Umatic. Then 1/2 with Beta being most popular. 44.1Khz fits mathematically with the video signal/ line rate.
The correct answer is FLAC. I really don't think there is merit to the argument that a FLAC will sound worse than a WAV. I don't buy the argument about power supply fluctuations. FLAC doesn't take much more CPU to play than a WAV, especially on relatively modern hardware. Honestly a high bitrate MP3 will be almost indistinguishable from the FLAC or WAV anyway. I wouldn't use ALAC because there's no benefit to that over FLAC and it locks you into the Apple ecosystem.
I have found always that EAC is the best ripper, I have also found that once you do a full install of EAC and perform all the tests and verify that you system(and drive) is 100% with EAC. Then other programs will also rip with this 100% which I found to be the case with Itunes. I did test it by comparing the wave files each program produced with EAC
I use FLAC stored on my NAS drive which I cast to my Chromecast Audio, plugged into my Cambridge audio DAC. I use Hifi Cast on Android to play the files and it's true gapless playback. The lights change accordingly with different sample rates on the DAC.
I thought to mention that there is an old file format called IFF that was used on the old Amiga computers back in the 80’s and 90’s, mostly for graphics, but also audio. If you look at the headers of AIFF and IFF, you will see that it is clearly built around the same system, hence it is not entirely correct to say that Apple made AIFF when Electronic Arts laid the foundation with IFF.
Bill Crane : Why ? is this EAC so good ? does it provide anything better ? Does it make 100% identical copies and other softwares can't ?? I mostly use the windows to rip a cds in the hard drive or CDBurnerXP and it's all good Currently stil using windows 7 64bit To be honest yesterday i've installed this EAC in my pc and except a few interesting features it has,, the sound quality is the same for uncompressed files , i mean i didn't notice something different for the better .
Secure ripping is the most important, as the CD was designed to conceal and not report reading errors. For Windows the best program for it is CUETools. It can verify already made copies against other submissions to its online database. Then you can re-rip the ones that do not match with a secure and slow reader like EAC. Upsampling is best done on the fly, with whatever output requirement you currently have, without inflating the file size on disk. WAV does have metadata, but it doesn't map quite well to music tags. Sonic Foundry regions and markers seem to be well supported. I had Nero write CD index points from regions that existed in the Wav file. Current players may stick an ID3 block into the WAV file for unlimited metadata. The RIFF format can serve as a container for anything. WAV is a good format for editing. Programs that support compressed files will either convert to WAV proxy beforehand, or be laggy as they do seeking in the compressed file on the fly. Back in the old days WAV could also be compressed with any CBR codec. That is bad because you can't see in the file manager the codec when all files have the same extension. In the Apple system this still happens, as ALAC and AAC will have the same extension. For de-emphasis, SoX works well. You can process the CD as a disk image to avoid clicks on track boundaries, remove DC offset and boost the level while in 32-bits accuracy. Old CDs are rather quiet, and theoretically there is a reduction in SNR while de-emphasizing.
Since we are discussing ripping CD's in this video, I thought I'd give a shout-out to the Linux users. "Whipper" is an excellent cli tool, and flacon is a good GUI tool for Linux (I use Arch) bit-perfect CD ripping. Nice alternative to Sound Juicer.
I've been ripping with Sound Juicer to FLAC for years. It works great, but I'm always open to trying new tools. I'll definitely check out Flacon. Thanks for the tip. Rubyripper is another one I've had good results with.
EAC to rip the CD to WAVE, TLH to convert WAVE to FLAC. Anyone who says they can hear the difference only thinks they can hear it! Don't listen to tech too much, just enjoy the music!
WAV files are composed of RIFF chunks, and they most certainly do provide for metadata. They also accommodate compressed audio data, although this capability is rarely used. Header, Format and place-holder chunks found at the top of the file are generally quite small, 8-32 bytes. As a rule, metadata chunks in any audio file format are placed at the end of the file, following all of the playable audio data, as they can be fairly large if they include cover art, etc.
Nice to know, thanks Marianne... but after I've clicked on "Properties" to see song (or metadata) info, it won't let me type anything into the boxes !!
i am sorry you might have a very very small delay on flac compared to wav , no jitter no power supply stuff, if you use flac. why the hell wont you play it with a computer that is able to play flac without any problem or 100 flacs at the same time for all that matters. the computing power to do so is rather low, especially nowadays. a phone could play multiple at one time
he needs to sell u $2000 power supplies; you cant get better power supply stabilty than with Li Ion batteries, but hey, he needs to sell you his toroidal custom transformers at gold price
There is not even anything processed. The "Audio Chip" in your gear has hardware acceleration for it. If aany it uses micro amps on the power supply. The power supply could not care less.
When I RIP cds on my Samsung laptop with the iTunes app, I use the Apple Lossless Encoder. The ripped cd sounds identical to the original and the disc will play in all players. For my phone, I use aac files. Nearly all the songs are cd quality.
ripped 10-50 cd's at 192k mp3 years ago. ripped 100's of cd's at 320k mp3 in the past 10 years. ripped 1000's of cd's in FLAC in the past few years. guess i will re-re-re-re-re-re-re-re-re-rip a few thousand cds in wave files. nothing like decades of ripping cds to play on other things, then playing the cd itself in a cd player.
I did the exact same thing, except no way I'll convert to wav. If I did, I would use a program like dBPoweramp to convert the FLAC to WAV (and thus lose the file tags). I can't imagine there's a difference in playback unless it's player related, I use Foobar2000.
I like XLD. You have to get past its "This will not work on a Mac" warning - this is false it works great and is independent from iTunes. It makes great FLAC files and is free
On Linux, I compressed my entire cd collection to FLAC using abcde when I moved countries a few years ago. I sold all my CDs to a second-hand store and no longer keep physical media. Abcde stands for Another Bloody CD Encoder. Once you have configured the app, you stick in the CD and type abcde in the terminal, and it does the rest.
Hearing the difference between AIFF and FLAC is the same discussion as saying there is an audible difference between a $5 and $500 USB cable. Many people claim they can hear a difference but there is no proof to back it up. Like Paul mentions the processing can introduce "something" but that has nothing to do with the format. The same goes for data cables. If bits arrive in the correct order, and all bits arrive on the other end, the signal was perfect. Whether that's over a cheap or expensive cable and yet people perceive a difference. It's the same as hooking up a new $200 power cable to the same outlet as before, with the same crappy wiring inside the walls and the same circuit breaker but yet, many claim to hear a difference. These perceived improvements (or snake oil, depending on the person you talk to) are scattered around in the audiophile community. In the end it doesn't matter I guess. If you think your DIY $20 speaker cables sound great on your overnight sensations DIY speakers, good for you. If you invested >$2000 in cabling alone but are sure it improved the soundstage, good. Whatever makes you happy about your setup.
Comment section here is interesting, and full of a lot of odd conclusions. I wonder how many people here have actually written de-compression code, and understand what exactly happens during compression and decompression. In a nutshell, the bits in a WAV file are in the correct order, and in theory can be streamed directly from the source to the amp. The bits in a compressed file and re-ordered, essentially, you have a couple of bits that say "play the next x-number of bits in the order they are in", then another couple bits that say "play the next x-number of bits y times," then another couple of bits that say "the next x-number of bits are exactly the same as the bit string you played z-times ago, so go back and use those bits." This process uses more processing power than people seem to think it does. The end product is exactly identical in every way to the WAV file, but the bit stream must be held in a buffer. So, yes, hardware and programming matter. If the player starts streaming a flac file, there potential for problems when it has to go back and re-use historical bits (what people seem to call power-supply jitter). If the bit stream is read into memory, then streamed, it will be exactly the same as a WAV provided there are no physical problems with the RAM chip and socket. The downside of this is you have to wait for the entire track to decode, which extremely fast from SDD, and pretty fast from HDD.
I use FLAC uncompressed it sounds great. I have compared it with WAV and hear no difference on a highly resolving system. I have heard FLAC is a better choice for long term storage .
I agree with keeping CDs as 16bit x 44.1kHz, but my reasoning is a little different--since that's the native CD format, you can't gain resolution that isn't in the source, but you can introduce additional errors recoding. We can always mess with resolution and sampling rates later! So I say the best thing to do is to keep it "as is", then if future DACs or software can do a better job of interpreting the signal to get back information lost in the recording/mastering process, you have the original digital stream to work from. It's been .flac for me for over 10 years now. I've ripped almost everything from CD now with artists tagged and album covers in the files. It's all on 2 machines at home, plus a cloud server. Don't want to lose it :-)
I agree with your reasoning. Also, I've got a PLEX server on my NAS and if i playback my music with an internet browser it plays back lossless no matter the lossless file format but if i playback ALAC or APE on my android mobile, it gets converted to lossy AAC, this is why i stick to FLAC
WAVE file is my choice. I record CDs with the music I like or sellected playlists. play WAVE files is my choice. - for some general music I like mp3 192 bitrate is enough to me
paul i use a 2012 imac 20"(2)Electro-Voice sentry 100 A studio speakers Crown Straight line preamp (2)cd recorder/players -Sony 10 band E-Q electro-voice interface BC EQ (2)tape cassette record/player (1)Crown 50w Power line amplifier
The real trick is how to properly rip a CD that was mastered with pre-emphasis and save it in a digital format that then always will play as originally intended.
I'd like to see any proof that anyone can hear AIFF sounding different from ALAC. Those folks you alluded to saying ALAC is too complex to get the best result need to be named, and we need to see their research.
Hi there PS Audio Team, please consider high pass filtering your audio on these vids. Some of us do listen to youtube on our systems :) and with a sub there is a constant low rumble in nearly all your videos.
One interesting thing about AIFF/ALAC is that if you rip stuff in ALAC and down the road decide you don't want to have to decompress it on the fly you can tell iTunes to decompress all your files and it's like you ripped them that way. No losses so no difference at all. If you're caching them to say an SSD and actually playing your music from that there's no need.
Newbie viewer; first-time comment... Enjoy watching your Q&A. I especially valued today's episode having to do with audio codec and sampling rate. Also wanted to say how much I enjoyed the episodes regarding building the new studio and unboxing of the Infinity Reference Speakers. Thanks again, Paul.
If the end result after processing sounds any different for any reason, then the problem is the DAC and filter. A proper DAC and filter should produce the same output regardless of what the PSU is doing as long as the PSU is operating within normal parameters. But as I've said before, there is no such thing as a perfect filter.
AIFF is actually "Audio Interchange File Format" en.wikipedia.org/wiki/Audio_Interchange_File_Format and is essentially a container for the actual data bits that represent the music. It was intended for uncompressed PCM and is mostly used for uncompressed PCM, so its a lossless format that retains all the original music quality. That said, AIFF behaves as a container and can be used to transport many different compressed audio files and each one can have different levels of compression. This is an additional reason why a few folks thought that they heard poor audio quality when listing to an AIFF file when in reality that file had been processed and compressed to a lower quality standard than the original uncompressed PCM file. Of course, where humans are involved there is always some sort of corruption of the original lofty goals and intentions. LOL Upsampling to 16/96 or any other sample rate does nothing to improve sound quality because you can't create sounds that weren't on on the CD to begin with. Of course playing back files through an audiophile grade audio system will sound better (with no changes at all) so I suspect that this is why some folks swear that that can hear clear improvements when none can actually be possible. Just play the bits without changing them on the best grade gear you can get and you're all set. BitPerfect is an outdated player that doesn't seem to perform any function anymore since iTunes doesn't change the bits (sample rate or dynamic range) any more. iTunes did change the bits in the past because we used to go into the Preferences pane and muck around with small rates there. That dates back to the old days where storage systems were tiny and putting an entire library of CDs on your personal computer would fill the drive yup completely. Obviously that is not a problem any more so that reasoning is moot. iTunes now handles FLAC files natively so that reasoning is also moot. In fact if you look in the Preferences pane of any reasonably modern Mac you'll see thither is no longer a setting to change the bit rate of any music when you import it, so whatever bits you get it will be the bits that go out. Save the $10 and out it towards some good music tracks instead.
@E. O. Well FLAC is still not an international standard and it still uses contested IP, so its not standard and can be shut down with a single lawsuit. That is why iTunes has Apple Lossless which is reliable and cannot be shut down.
First time i've watched some of these video's from here in the UK and i have to say that I am very impressed by the methodical way that Paul explains everything. Peoper old school with no bull shite! Will bewatching more regularly as I'm very passionate about good sound quality. Especially at Home and in my Car.
Hahahaha... you are serious that you care about audio quality in your car? I care about sound quality in the shower, in underground tunnels and when traveling by train.
I have been ripping CD's using the Mac Finder and getting the AIFF files instead of using an actual ripper from iTunes or some other player. Amongst the agreements and disagreements smoke over AIFF vs FLAC I figured storage is cheap so just go with AIFF. Furthermore, compared to WAV I can store meta data and album art in each track and customize that.
One of the best questions and answers around. Still, its a standard answer in general though. Increasing any technical levels will not make it better than the original recording. If you have 5 dollars in your pocket, putting into into a bigger pocket will not make it 10 dollars. Also, It might just get lost.
Storage is cheap, Paul - even for a cheap bastard like me. I use AIFF for archiving but I'll listen to ANYTHING : unlike the snobs, I don't turn up my nose at the 320K Spotify Premium downloads on my laptop. At 60, the worst link in the chain is always going to be my ears - as much as it pains me to admit that - and 30 seconds with a frequency sweep confirms that I just cant hear what I could when I did the initial hearing tests for the Army as a youngster. IMO, the best way to rip CDs is with Exact Audio Copy - ymmv. Thanks for the video.
I just ripped all my moms cds to 128kbps MP3 files and I'm wondering what the difference is? Playing the song from the cd in windows media player sounds the same to me as playing the song from the rip (in WMP not grove music)
there are differences, yes. 128 or WAV yes, and it depends the speakers or component audio system you have to play the audio. my sellected playlists with my favourite music commonly is in CDs I record - sometimes the original is mp3, I convert ot to WAVE, I make some improvements if it was necessary and I get my personal CD witth good quality. sometimes the USB with mp3 is enough and I enjoy it too.
@@Ephemeral2023 some songs or videos from youtube (and other different formats I could have) are a mess with poor quality. I have songs since Audiogalaxy or Napster times, even some iTunes that I've purchased need some "restoration": to apply some filters and saving them as WAV files.
@@Ephemeral2023 in 40 years I've listened music, MP3, WAV, AAC, cassette, LP, Minidisc, I enjoy my music. My mother and my neighbor enjoy this or those and their music, but there is a difference between listen in Mono AM RADIO mode than FM Stereo, _it is better to enjoy more_ commonly it is said as _it sounds better_ songs I ENJOY sound better, a few times I save the file as MP3. similar to the movie on TV, "the important is to enjoy the movie" yes, but with a better high quality format is better than VHS. 🤷✈🔊
@@Ephemeral2023 no. really I don't. when I heard the difference between a youtube file and a WAV file after enhance the waveform, and sometimes cleaning up the original file (even from CD tracks), I get a cleaner and better audio file, and I see majoroty of tracks are clipped. of course, I like the process, I have the time and tools to do it with music I'm intereted and I enjoy, because I see the difference between a pig with lipstick and without it. form yuour own point of view to you, it is useless, because you aren't interested, you don't have the ABCD and you don't need it. if you have a different way to listen whatever, I like mine.
Though certainly not an expert (I don't even rise to the rank of novice or beginner), I can think of a reason that 24-bit, 96k sample rate could be helpful. Think of it like video frame rates. 24 (film) vs 30 (old video), vs 60 vs 120 frames per second. Suppose the original recording was at 24 FPS for film. If you watch that 24 FPS video frame-by-frame, you see noticeable jump between the frames. While you can't get any NEW information, you can do a linear interpolation between the 2 samples to have a smoother transition between the frames. If you're up-sampling from 24 to 120 FPS, you now have 4 additional frames that have been interpolated between the 2 reference frames. Then, viewing the 120 FPS video frame-by-frame, there's less jump between frames, so it would look smoother. I imagine it could be the same with audio. Finally, with advanced signal processing, not only could you do linear interpolation, but you could also consider the samples before and after the ones between which you're interpolating, and maybe fill in the missing samples based on the preceding and following flow of the music.
Interesting to see the comments that decompressing FLAC only uses a small processing overhead, and that a phone could deal with loads of tracks at once. Also disappointed that Apple will only allow it to run in the OS on iPhone 7 and above as it drains the battery too fast on a weaker chip.
I can play FLACs on my 6s+ with FE File Explorer, I also use it to load my mp3's, FLACs, and MKV/MP4 videos from my NAS to the phone/iPad for travelling
@@Snowwie88 I can\t tell a difference between FLAC and 320 kbps MP3. Still, idiot as I am, I stick to FLAC in case I ever buy equipment good enough to actually hear a difference.
dBPoweramp is the program to use to rip your CDs or batch convert your digital files of any kind. Exact Audio Copy is OK, but isn't quite as good at giving you exceptional "Bit Perfect" results as consistently as dBPa is able to .
What happened to CDA? Isn't the format of the data on an audio CD recorded as CDA as in compact disc audio format? With the size off modern drives being so huge what is the matter with just recording it as a CDA file? That way the bits are exactly the same as on the compact disc, right?
Did 0:40 make you curious, So then end up touching with your finger the inner grid even after the button was released and find that it hurts or at least suprisingly gave enough to be uncomfortabke enough to not want to do that ever again if avoidable/if not an inadvertent thing to have happen again at least not on purpose... *I did*
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I normally play sound system on mac at 16/44 no dolby and when I use 24 bit files I use my mackie firewire mixer and audio technica headphones. ... but bitperfect sounds like a neat solution so you don't have to keep switching .
4:20 I would love to hear an elaboration on this concept. Especially, "a revealing system." I find this incredibly interesting. Thanks for the information :)
Do you think that different programs sound different (assuming it's not doing any processing to the file)? I've heard people say that they do but I haven't been able to hear a difference in my own testing and I don't see how it could if the program is just feeding identical data into a buffer for the audio driver. I assume that there isn't any sort of clocking involved at that stage. I can see how using a different driver could make a difference but not a difference playback program.
I'm not talking about resampling or any other processing. Just playing a file bit transparently. Some people say that some players like JRiver still sound better.
On a computer @ 192kbps if you have a lot of songs I find I get more on my Sony 64gb device at that ripping bit rate, and I recommend using the SonyMediaGo app and with all that Sony give in their sound settings on my device there is not a massive loss in detail by ripping at a lower bit rate👍😃👍 Flac this Flac that blah blah better sound etc etc but like I said before at that vastly high bit rate your MP3 player is going to fill up pretty quickly if you like a lot of songs, so beware go down to 320kbps or do as I do 192 or a happy medium at 256 the choice is yours 👍
Rip to wav using dBpoweramp. Then do the conversion to Flac after. That way jitter can not enter due to the simultaneous demand of the psu and computer processor. I also close all apps and background tasks. Use the securerip feature (which checks your rip against there database of rips) for errors and if there are any it re-rips multiple passes at lower drive speeds till to improve it. Not add cd's are flawless and it isn't long before you have the odd scratch. Personally what I do if I cannot rip at 99% or greater accuracy I will torrent the audio cd. ( I own the album on at least 2 formats).
Many folks simply use their Windows Media Player to rip and archive CD's to 'lossless' form. They choose 'highest quality' to rip to WMA lossless. I do that as well, especially when I sync my portable FiiO player to my PC, in order to transfer lossless files to it. What are the advantages or disadvantages to simply using the Windows Media Player for ripping CD's in lossless?
Hi Jeremy. Sometime, on some of my PC's, it does! But lately, I have been using my laptop with Win 7, and The sync only takes a few seconds. On my other Win 7 and XP machines, it takes much longer, and sometimes doesn't even sync up at all. I also use an FiiO X1, and love it! That said, I may just try 'Media Monkey'. Thanks.
If there were a few bits missing in the copy you may not notice it, but it does in fact happen, especially if the disc is damaged or deteriorating. I had several that had to be ripped several times to pass the verification check, and several that were damaged and never did, one of which I had to repair the dropouts. I mistakenly said EAC before but it's it's actually AccurateRip that Foobar2000 uses now, though I think I had to install EAC to get that feature. See also www.accuraterip.com/ DVD players in computers (maybe not all, I don't know) can't even play CDs slow, so it's even a bigger problem now. I'm pretty sure I switched to AccurateRip/EAC when I eventually noticed a dropout in one of my CD tracks.
Err..EAC is NOT necessarily the best, but does tend to get the most detail from the disc. Depending on how that info is outputted will determine whether it soundstoo digitally harsh, for me, when streaming 16/44 EAC wavs, it totally did. BUT, for re-burning CD's all that top end information is a hella good, as the process itself will incur some loss just by the burning process being largely mechanical (but huge gains else where). For ripping to stream 16/44, I personally only use XLD on MAC, much more forgiving on the ears, on my system in my room. Try it, they sound waay different.
Ye, and one thing that wasn't mentioned. Transporting wav files is slower due to the size. So id someone has slow discs, bad wfi etc. It takes a bit more time. So could cause stutters. Also, it is twice as expensive to store!
In other words if the source is analog, sample @ 24-96. But if the source is 16-44.1 digital, just copy in the native format. It's like in photography, they sell us numbers. Who needs a 45X zoom? Not me....
should I keep ripping my CDs to Apple Lossless or switch to AIFF? I recently tested both and can hear a subtle difference, in drums and vocals, but not sure if I'm tripping
I use AUDACITY for any/all of my CD recordings and then add DOLBY C from my cassettes for that professional sound onto the CD and it sounds great and all you do is take the large end from the cable and plug into the hard,drive and then take the 3mm jack and plug it into the headphones output and then record your CD from the computer.The DOLBY B or C makes the hiss much lower when you play it back.
Uh, no. Dolby NR is not a single-ended noise reduction process. It is only applicable to recordings that were originally made using the process at the recording end and then during playback. Otherwise, the equalization will be totally whacked. The result would be similar to playing an LP record without RIAA equalization in the phono preamp. BTW, Dolby NR is obsolete and no longer licensed by the company. No contemporary commercial tape recording can be made with the process and therefore should never be played back with Dolby NR. If you have an old tape deck that incorporated Dolby B/C then you could still record with it turned on and then play it back with it also engaged.
Use AIF most of the times and other files ,Flac ,high resolution some DSD and use ROON which is by far the best software for music and can play any files and is regularly up-grated (best investment I ever did) As external hard disks became reasonable in price I do not see the need to use Flac or ALAC .
All you need is a good burning application, like Burning Studio,to burn files or Rip CDS This is the program I use to burn files or Rip CDs.Even burn movies
personally i use a program called audiograbber and rip the files in .wav . it's an older program but performs in a very efficient manner. it will also compress into .mp3 but i don't use that feature nor the"normalize" function as i prefer to store and playback the way that the engineer and artist created the stored content.
There is such thing as "non destructive replay gain" where the gain values are stored in metadata on a track and album level. This has no effect on the encoded files audio. You can turn of this "replay gain" function within your playback/decoder software.
Audiograbber is all I use myself. I paid for it years before it was turned into a free program and development stopped. It still works great, basic and easy to use. It is really fast on newer computers, compared to the old top of the line 286, 386 and 486 computers when it started.
With my system set for foobar2000 on-the-fly conversion of audio files to 512DSD, I have found that DBpoweramp CD Ripper ripping to flac file setting 'no compression' produces a file which is identical to the sound of the CD used for the rip: any setting other than 'uncompressed' produces a less-resolved sound. I realize that the cost is increased bytes of storage, but my priority is the best possible sound when making or listening to CDs or audio files: drive.google.com/drive/folders/1C1w4HcZuThrTxImadujOdftffK3s_2K3?usp=sharing
I used to chase flies around furiously. And then I thought "All creatures have a right to be" and I ignored them. The odd thing is now they RARELY give me any trouble at all...
There is no difference between WAV, FLAC, ALAC. They are audibly equivalent storage of uncompressed PCM data. If you hear a difference, it is confirmation bias. Use a tool called Audio Diff Maker, which shows the difference between any two audio files. There may be differences in playback hardware, but not the files themselves. For example, some DAC's clip at 0dBFS, others do not. It is DAC chip dependent. There is also inter-sample clipping. That is, clipping above 0dBFS BETWEEN discrete sample points. Plus, this phenomenon would be exhibited no matter what file format was being played. So, there can be some reasons that HARDWARE sounds different, but the files themselves are identical in every aspect.
You carve the wood by tool or with the machine. The result will be the same at first sight... Proces by handtool is slower more precise and silent, machine is much faster and noisy but not so precise since it is designed to use its power for speed, not precisions and fine carvings. Most of the people will be satisfied with the former but not everyone. So, your software shows only the curve of the sound because software actually cannot show music - only ear can. So I don't know what you and people like you try to prove especially with software. Look at. perhaps, ringing that Bob Stuart claims that affect the sound. Where is that in your software. Pre and post ringing when you turn digital bits into analog electrical wave??? I had discussion recently with the person that brought me into the audio and he told me that crossover doesn't pass the signal higher or lower than cut off freq. at all and I didn't agree and ask him to explain that from technical point of view and what are the first, second and third order crossovers? He had a misconception based upon common snese. My point is that engineers turn theory into product with the topology, chips, capacitors and millions of parts and I'm not nearly capable to put that in question with simple high school physic and "common sense". I can choose to believe or not to believe. Or I can try and decide. Software is redundant. And if you're so sure about your standpoint you should be happy that you're able to save some money without any loss. I suppose you're all altruistic and thank you for that.
Freekwo 777 The software actually does show the "music." Music is only the summed collection of sine waves at different frequencies and phase at the same relative time. There is no difference between the FFT impulse and the deconvoluted frequency response. They are both sides of the same coin. Frequency, Phase, Time, are all the same thing looked from different angles. They are easily represented and these representations are used by engineers every day to build the products you love and admire. As for your friend, he may have been a prime example of someone misguided by wrong info or not understanding what was told to him. I see it everywhere. Good luck in your quest.
Maybe you're not seeing it but you have just made the same mistake as my friend did but on the higher level of knowledge. Music simply cannot be seen, we sense music by ears so your fundamental mistake is more than obvious. When we talking music here we're talking about reproduction and the whole "Hi-fi" means high fidelity of reproduction. Maybe you should go to the concert with you cell phone and record the concert and suggest everyone to the same instead buying pro records on cd, vinyl because software shows the same curve and there is somekind of bias. Maybe Hollywood should do the same with video since some software would certainly show the colour scale the same. Or maybe the old classic movies are not good enough because average cell phone can do the better job nowdays. You're argument has completly miss the point and I'm glad that I stepped in this discussion because there are so many people that confusing other people with self made philosophy without any praxis and now we know that software shows music. Thank you for that fact. fortunately it's only a link in the audio chain and certainly isn't music since I never heard the curve.
Freekwo 777 Ok. Now I am getting to the point where it is hard to understand you. If music cannot be seen, measured, or quantified, then all the audio designers, engineers and sound techs, mixers, masters, and performers should just throw away all test gear and design by ear? Your statement bears nothing in fact, only subjectivism. History has taught us that subjectivism, "feels" if you will, is always wrong. Certainly music has an emotional element as we connect to music emotionally. However, we can certainly see and calculate waveforms, convolve and deconvolve impulse responses and SEE the outcome. By comparison, we can also SEE the effects of change. Why do you suppose EVERY designer of audio gear uses TOOLS, save for the layperson designing crap in their garage with a multimeter and soundcard "analyzer"? Dispell with mythology my friend.
And he's wrong about WAV metadata. As a derivative of RIFF, WAV files can be tagged with metadata in the INFO chunk. In addition, WAV files can embed any kind of metadata, including but not limited to Extensible Metadata Platform (XMP) data or ID3 tags in extra chunks. Applications may not handle this extra information or may expect to see it in a particular place. Although the RIFF specification requires that applications ignore chunks they do not recognize, some applications are confused by additional chunks. Video's junk.
What was that again with adding random bits for dithering? And that Nyquist hesitations? Nevermind that "flac could sound different than wav". Everyone can have a bad day. I'd just not post on one of those days ;)
@Mike P if it really is unaltered. I.e. qobuz vs Tidal is interresting. Both say lossless. Qobuz is true lossless and Tidal mqa has been analysed a lot and they add some loudness effects and certainly not lossless. I don't know how apple will deliver yet. I am on qobuz.
I like to rip CDs onto FLAC (with dbpoweramp) or ALAC (with iTunes) because they support metadata very well. Of course with 16/44.1 as CD standard does, I don't like something upsampled. If I want to hear the 24-bit music I just open my Qobuz or Tidal app
I would say most records would be fine sampled at 16 bits. There are some records that actually have a wider dynamic range than will fit within the old CD digital standard however. yes the noise floor on the record can be easily heard during the quiet passages, but so can the music. Recording to 16-bit, setting the peak of the recording to match to the maximum level of the digital standard, the quiet passeges drop to dugital silence with an periodic soybd here and there as the music and surface noise raises up to the lowest level allowed with 16-bits. I do not know what the dynamic range is with 24-bits... Another thing to consider is if you plan to do any processing to the audio after recording a record to digital, is that processing can negatively alter the audio very quickly with the low resolution of 16-bit. 24-bit allows the audio to be processed with little errors created while processing with digital filters, equalizers, noise processors, or even simple digital volume adjustments.
One possible advantage of ripping vinyl with a higher bit rate would be the increased headroom before aliasing. I've transcribed over 600 LP's to digital and use the standard 44.1 KHz/16-bit CD format. The main issue I had to deal with is that it can be very time consuming to preview every second of an LP to determine the maximum audio amplitude and set the digital recording device to the highest possible level (for the best resolution) without clipping (anything over 0 dB), which ruins a digital recording. My solution was to try to quickly find the loudest passages and set the recording level appropriately (for maximum resolution) but use a soft-knee limiter with look-ahead delay just ahead of the A/D conversion to knock down any unexpected peaks that would cause clipping. I use a tc electronic Finalizer Express (A/D/A converter) which has the limiter on the analog input. Absent a limiting function on your A/D converter you could just buy an analog limiter and place it ahead of the converter and then digitally record at the sampling rate you use for playback.
I have ripped my cd collection in wav by pc equipment and stored in my home cloud. Knowing that the rip quality cannot be "perfect" due to poor jitter and other factors (electrical/electronic noises) generated in the pc system during the process of ripping. Anyway, just enjoy the cloud music collection streamed by my two oppo players and Chromecast audio. I find wav format is "more musical", just personal opinion.
If you system is having difficulty decoding FLAC, you have an abacus for a CPU.
Furthermore, if your abacus processor taxes your power supply you have another problem.
My decoding abacus uses Mpingo beads, thank you very much. Much more analogue sounding.
@@cjsvega Do not use an Intel Celeron CPU, they are junk mainly for office use, like documents, spreadsheets, etc, you need a Pentium D or Pentium 4, Dual Core ideally a 2.0GHZ CPU or higher, or an i7 quadcore CPU such as i7 920 CPU.
@@speedyboishan87 no you don't, a music server playing FLAC ,wav or even mp3 will run just fine on just about any PC ever made to run Windows 95.
and that's prehistoric technology, but would still work.
winamp was free music server software thats still beats newer aps.
Or apple
WOW I'M FAMOUS :) !!!..this is my request, thanks Paul, if I haven't thanked you already.
Thanks for asking him
What a pleasure to watch informative videos presented by someone mature who's not trying to impress or be cool. Just straight down the line engineering. You are the duck's nuts of TH-cam Paul 👏
Paul has got to be one of the most engaged CEO's I've ever seen. What a great place PS Audio must be to work.
seems a very nice guy
Thanks! I hope you have the chance to visit someday.
paul has the best job in the world.
Michael B. Paul is generally not at his desk.
sädle he didnt know if his chippäir amp wöz 200? ör 2k. well was ja maybe it was 2k 4k 42 ?
but almost 5050 it was an extra diggit v$v
1:15 nö aströhöbbyy well nö ge xD
I intensely stare at the cd for about an hour and then write 1s and 0s in notepad++, save, and rename to wav.
(Edit:2020)
During the lockdown i have mastered the art of engraving CD by hand. I have now achieved replica quality of the original content. If the lockdowns get extended i might go so far to manually magnetize hard drive platters in order to achieve faultless source migration.
P. S. You people are one helluva community
Of course, I expect that there is a checksum for verification ? I'd really hate to end up with a dud copy of my Barry Manilow catalog.
@@arthurwatts1680 LOL, you idiots!
@@arthurwatts1680
WAV files do not include checksums (beyond that provided transparently by the media hardware such as disk sector CRCs), but when they are encoded for transfer to an audio CD, error detection/correction data is added.
@@marianneoelund2940 Thanks for the clarification, Marianne, but I think you'll find that Mr Jerkovic (!) and myself were just trying to inject a little humor into what is a very dry topic. There was a time when I agonized over what EAC was doing with my CD rips but it's been 6 years since I've purchased a CD so its all a bit beyond me now. I know - flat-earthers and all that - but it is what it is. MQA and DSD are equally irrelevant in my little world, fwiw, but I realise that the world doesnt revolve around moi.
@@arthurwatts1680
I appreciate that. But I often take opportunities like this to post trivia which might be interesting to someone.
The lack of checksums or CRCs in most audio files makes them rather easy to hack or modify at the bit level - something I've been taking advantage of recently.
As a computer scientist i can say flac and wav are identical... if you are worried about the size reduction, dont be.
As for the possibility of sounding worse due to the extra processing to uncompress, i see your reasonining, but i see that as an extremely rare case! If that happened to you, your dac has a really really major flaw, because decompressing the flac is super simple stuff, just throw away that dac!
What everyone here seems to be missing is the fact that even Paul said the decoded bits are identical. In other words, he agrees with most of these arguments in the comments. I believe the confusion comes from the fact that he is often digressing to address the entire signal path from the media to the speaker output rather than the differences in the file format.
Even though the decoded bits are identical, it is easily possible for the final analog waveform to vary noticeably. The differences people can hear are not due to changes in the bits. Like everyone (including Paul) keeps saying, the bits are identical. The difference in sound comes from other factors. First, there can be minor changes in the amount of time between the bits during the playback. (Not decoding.) These tiny differences actually DO alter the final analog output signal which is created from the bit stream. An extremely accurate waveform comparison between the analog output signals being sent to the amplifier would show those differences. When the width of the pulse changes, the waveform changes. That's the entire premise of Pulse Width Modulation.
Another contributing factor is the hardware and software involved in processing the identical bits. Most audio circuitry does some level of filtration at various stages during the conversion process back to analog. These filters assume that the amount of time between the bits conforms to the sample rate perfectly, which would (in theory) produce an exact replica of the original waveform. This is often not the case. Digital filtration, in particular, is highly timing dependent. If the spacing changes, the effect of the filter will also change. Analog filtration can also change the sound since it is based on the controlled attenuation effect of capacitors and inductors responding to varying frequencies and amplitude within the signal. Then, there is the fact that the actual values of electronic components vary randomly within a specific tolerance range, which will also affect the output.
I'm quite certain that there are myriad other factors along the signal path which can slightly alter the analog output in a way that affects the fidelity of the output waveform to the originally recorded waveform. So, the bottom line is that, although the bits are precisely identical, the output can indeed be physically (and even audibly) different due to the influence of other factors. What really matters is whether you enjoy what you hear, and that is purely subjective. So, why pick on others who love music as much as we do?
Thank you, but having said all this, surely there is a process which would ensure all these errant bits remain in or can be returned to their correct positions ?
I haven’t had a problem decoding flac files since the early 2000s.
I Rockbox'd my old mp3 players, in order for it to play FLAC & other lossless format. The weak processors in these device most certainly decode FLAC in real time, even huge merged audiobook files. The open source firmware let's user peek the computation as the file plays. One has to be running something ancient to not practically decode level 8 FLAC
44.1Khz was selected because originally the audio was processed in one unit and storage was a video tape recorder. Originally 3/4" Umatic. Then 1/2 with Beta being most popular. 44.1Khz fits mathematically with the video signal/ line rate.
The correct answer is FLAC. I really don't think there is merit to the argument that a FLAC will sound worse than a WAV. I don't buy the argument about power supply fluctuations. FLAC doesn't take much more CPU to play than a WAV, especially on relatively modern hardware. Honestly a high bitrate MP3 will be almost indistinguishable from the FLAC or WAV anyway. I wouldn't use ALAC because there's no benefit to that over FLAC and it locks you into the Apple ecosystem.
Accurate.
Putting aside power, there are other factors involved when "on the fly" decoding happens. It also depends on how hardware/amplifier perceives it.
Only flac level 0
Don't rip with itunes, argh... Rip with EAC or foobar into 16/44.
why?
@Alex X i dont get it
@@bnhintz You want to use a secure rip program for greater accuracy.
@@Slammy555 thank you
I have found always that EAC is the best ripper, I have also found that once you do a full install of EAC and perform all the tests and verify that you system(and drive) is 100% with EAC. Then other programs will also rip with this 100% which I found to be the case with Itunes. I did test it by comparing the wave files each program produced with EAC
I use FLAC stored on my NAS drive which I cast to my Chromecast Audio, plugged into my Cambridge audio DAC. I use Hifi Cast on Android to play the files and it's true gapless playback. The lights change accordingly with different sample rates on the DAC.
I thought to mention that there is an old file format called IFF that was used on the old Amiga computers back in the 80’s and 90’s, mostly for graphics, but also audio. If you look at the headers of AIFF and IFF, you will see that it is clearly built around the same system, hence it is not entirely correct to say that Apple made AIFF when Electronic Arts laid the foundation with IFF.
roygalaasen
Sounds iffy ;-D
Amiga is still alive, ! and steadily being upgraded through its trap door, or add on processor interface.
Only problem vaporising inspects is when the cooked bits land in your coffee and you don't notice.
For PCs use Exact Audio Copy to rip CDs.
Bill Crane does exact audio copy rip cds fast?
EAC is the best tool for rip accurately
And it can convert to FLAC also when ripping.
Bill Crane :
Why ? is this EAC so good ?
does it provide anything better ?
Does it make 100% identical copies and other softwares can't ??
I mostly use the windows to rip a cds in the hard drive or CDBurnerXP and it's all good
Currently stil using windows 7 64bit
To be honest yesterday i've installed this EAC in my pc and except a few interesting features it has,,
the sound quality is the same for uncompressed files ,
i mean i didn't notice something different for the better .
@@Residentombraider1000 Yes.
Secure ripping is the most important, as the CD was designed to conceal and not report reading errors. For Windows the best program for it is CUETools. It can verify already made copies against other submissions to its online database. Then you can re-rip the ones that do not match with a secure and slow reader like EAC.
Upsampling is best done on the fly, with whatever output requirement you currently have, without inflating the file size on disk.
WAV does have metadata, but it doesn't map quite well to music tags. Sonic Foundry regions and markers seem to be well supported. I had Nero write CD index points from regions that existed in the Wav file. Current players may stick an ID3 block into the WAV file for unlimited metadata. The RIFF format can serve as a container for anything. WAV is a good format for editing. Programs that support compressed files will either convert to WAV proxy beforehand, or be laggy as they do seeking in the compressed file on the fly.
Back in the old days WAV could also be compressed with any CBR codec. That is bad because you can't see in the file manager the codec when all files have the same extension. In the Apple system this still happens, as ALAC and AAC will have the same extension.
For de-emphasis, SoX works well. You can process the CD as a disk image to avoid clicks on track boundaries, remove DC offset and boost the level while in 32-bits accuracy. Old CDs are rather quiet, and theoretically there is a reduction in SNR while de-emphasizing.
Que ??
EAC wouldn’t run on my Linux desktop. It kept crashing.
Since we are discussing ripping CD's in this video, I thought I'd give a shout-out to the Linux users. "Whipper" is an excellent cli tool, and flacon is a good GUI tool for Linux (I use Arch) bit-perfect CD ripping. Nice alternative to Sound Juicer.
Yay for Linux............
I've been ripping with Sound Juicer to FLAC for years. It works great, but I'm always open to trying new tools. I'll definitely check out Flacon. Thanks for the tip. Rubyripper is another one I've had good results with.
Paul, I can not honestly count the number of free tweeks & improvements you’ve supplied me with! A big Thank You! Sharing is caring.
EAC to rip the CD to WAVE, TLH to convert WAVE to FLAC. Anyone who says they can hear the difference only thinks they can hear it! Don't listen to tech too much, just enjoy the music!
Why not EAC to FLAC? It's the same.
WAV files are composed of RIFF chunks, and they most certainly do provide for metadata. They also accommodate compressed audio data, although this capability is rarely used. Header, Format and place-holder chunks found at the top of the file are generally quite small, 8-32 bytes.
As a rule, metadata chunks in any audio file format are placed at the end of the file, following all of the playable audio data, as they can be fairly large if they include cover art, etc.
Nice to know, thanks Marianne... but after I've clicked on "Properties" to see song (or metadata) info, it won't let me type anything into the boxes !!
i am sorry you might have a very very small delay on flac compared to wav , no jitter no power supply stuff, if you use flac. why the hell wont you play it with a computer that is able to play flac without any problem or 100 flacs at the same time for all that matters. the computing power to do so is rather low, especially nowadays. a phone could play multiple at one time
he needs to sell u $2000 power supplies; you cant get better power supply stabilty than with Li Ion batteries, but hey, he needs to sell you his toroidal custom transformers at gold price
There is not even anything processed.
The "Audio Chip" in your gear has hardware acceleration for it. If aany it uses micro amps on the power supply. The power supply could not care less.
When I RIP cds on my Samsung laptop with the iTunes app, I use the Apple Lossless Encoder. The ripped cd sounds identical to the original and the disc will play in all players. For my phone, I use aac files. Nearly all the songs are cd quality.
I love mastering then exporting in pcm 16 bit 44khz uncompressed loseless 💿 it plays on almost anything from 1980 to now and sounds great :)
ripped 10-50 cd's at 192k mp3 years ago.
ripped 100's of cd's at 320k mp3 in the past 10 years.
ripped 1000's of cd's in FLAC in the past few years.
guess i will re-re-re-re-re-re-re-re-re-rip a few thousand cds in wave files.
nothing like decades of ripping cds to play on other things, then playing the cd itself in a cd player.
I did the exact same thing, except no way I'll convert to wav. If I did, I would use a program like dBPoweramp to convert the FLAC to WAV (and thus lose the file tags). I can't imagine there's a difference in playback unless it's player related, I use Foobar2000.
No need to re-rip any, just decode your flac back to wav.
I like XLD. You have to get past its "This will not work on a Mac" warning - this is false it works great and is independent from iTunes. It makes great FLAC files and is free
On Linux, I compressed my entire cd collection to FLAC using abcde when I moved countries a few years ago. I sold all my CDs to a second-hand store and no longer keep physical media. Abcde stands for Another Bloody CD Encoder. Once you have configured the app, you stick in the CD and type abcde in the terminal, and it does the rest.
Not good, now you no longer have ownership rights and essentially have bootlegged music. If asked. How would you prove you bought the music?
Hearing the difference between AIFF and FLAC is the same discussion as saying there is an audible difference between a $5 and $500 USB cable. Many people claim they can hear a difference but there is no proof to back it up. Like Paul mentions the processing can introduce "something" but that has nothing to do with the format. The same goes for data cables. If bits arrive in the correct order, and all bits arrive on the other end, the signal was perfect. Whether that's over a cheap or expensive cable and yet people perceive a difference.
It's the same as hooking up a new $200 power cable to the same outlet as before, with the same crappy wiring inside the walls and the same circuit breaker but yet, many claim to hear a difference. These perceived improvements (or snake oil, depending on the person you talk to) are scattered around in the audiophile community. In the end it doesn't matter I guess. If you think your DIY $20 speaker cables sound great on your overnight sensations DIY speakers, good for you. If you invested >$2000 in cabling alone but are sure it improved the soundstage, good. Whatever makes you happy about your setup.
isolating your equipment from the grid and building wiring does make a difference. Not only in sound quality but it makes your equipment last longer.
You take forever to give a final answer
Comment section here is interesting, and full of a lot of odd conclusions.
I wonder how many people here have actually written de-compression code, and understand what exactly happens during compression and decompression.
In a nutshell, the bits in a WAV file are in the correct order, and in theory can be streamed directly from the source to the amp. The bits in a compressed file and re-ordered, essentially, you have a couple of bits that say "play the next x-number of bits in the order they are in", then another couple bits that say "play the next x-number of bits y times," then another couple of bits that say "the next x-number of bits are exactly the same as the bit string you played z-times ago, so go back and use those bits." This process uses more processing power than people seem to think it does.
The end product is exactly identical in every way to the WAV file, but the bit stream must be held in a buffer. So, yes, hardware and programming matter. If the player starts streaming a flac file, there potential for problems when it has to go back and re-use historical bits (what people seem to call power-supply jitter). If the bit stream is read into memory, then streamed, it will be exactly the same as a WAV provided there are no physical problems with the RAM chip and socket. The downside of this is you have to wait for the entire track to decode, which extremely fast from SDD, and pretty fast from HDD.
I use FLAC uncompressed it sounds great. I have compared it with WAV and hear no difference on a highly resolving system. I have heard FLAC is a better choice for long term storage .
I agree with keeping CDs as 16bit x 44.1kHz, but my reasoning is a little different--since that's the native CD format, you can't gain resolution that isn't in the source, but you can introduce additional errors recoding. We can always mess with resolution and sampling rates later!
So I say the best thing to do is to keep it "as is", then if future DACs or software can do a better job of interpreting the signal to get back information lost in the recording/mastering process, you have the original digital stream to work from.
It's been .flac for me for over 10 years now. I've ripped almost everything from CD now with artists tagged and album covers in the files. It's all on 2 machines at home, plus a cloud server. Don't want to lose it :-)
I agree with your reasoning. Also, I've got a PLEX server on my NAS and if i playback my music with an internet browser it plays back lossless no matter the lossless file format but if i playback ALAC or APE on my android mobile, it gets converted to lossy AAC, this is why i stick to FLAC
WAVE file is my choice. I record CDs with the music I like or sellected playlists.
play WAVE files is my choice. - for some general music I like mp3 192 bitrate is enough to me
I put on protective eyewear (use a full-face shield if you've got one) and gloves and use 2 pair of Vice-Grips.
paul i use a 2012 imac 20"(2)Electro-Voice sentry 100 A studio speakers Crown Straight line preamp (2)cd recorder/players -Sony 10 band E-Q electro-voice interface BC EQ (2)tape cassette record/player (1)Crown 50w Power line amplifier
Great summary of formats... much appreciated
There's a bottle of smart water on the desk to help with the math . :)
The real trick is how to properly rip a CD that was mastered with pre-emphasis and save it in a digital format that then always will play as originally intended.
ooh wee!
Rip the image.
Use Imgburn or PowerISO and make an image of the disc. No ripping needed.
I'd like to see any proof that anyone can hear AIFF sounding different from ALAC. Those folks you alluded to saying ALAC is too complex to get the best result need to be named, and we need to see their research.
Hi there PS Audio Team, please consider high pass filtering your audio on these vids. Some of us do listen to youtube on our systems :) and with a sub there is a constant low rumble in nearly all your videos.
One interesting thing about AIFF/ALAC is that if you rip stuff in ALAC and down the road decide you don't want to have to decompress it on the fly you can tell iTunes to decompress all your files and it's like you ripped them that way. No losses so no difference at all. If you're caching them to say an SSD and actually playing your music from that there's no need.
Newbie viewer; first-time comment... Enjoy watching your Q&A. I especially valued today's episode having to do with audio codec and sampling rate. Also wanted to say how much I enjoyed the episodes regarding building the new studio and unboxing of the Infinity Reference Speakers. Thanks again, Paul.
If the end result after processing sounds any different for any reason, then the problem is the DAC and filter. A proper DAC and filter should produce the same output regardless of what the PSU is doing as long as the PSU is operating within normal parameters.
But as I've said before, there is no such thing as a perfect filter.
@Taco How do you listen to your copies?
@Taco how do you think you hear the music? you need a DAC.
What do you think of ATRAC?
Excellent yet easy to understand answer. The kind I was looking for a long time without sounding too technical.
AIFF is actually "Audio Interchange File Format" en.wikipedia.org/wiki/Audio_Interchange_File_Format and is essentially a container for the actual data bits that represent the music. It was intended for uncompressed PCM and is mostly used for uncompressed PCM, so its a lossless format that retains all the original music quality. That said, AIFF behaves as a container and can be used to transport many different compressed audio files and each one can have different levels of compression. This is an additional reason why a few folks thought that they heard poor audio quality when listing to an AIFF file when in reality that file had been processed and compressed to a lower quality standard than the original uncompressed PCM file. Of course, where humans are involved there is always some sort of corruption of the original lofty goals and intentions. LOL
Upsampling to 16/96 or any other sample rate does nothing to improve sound quality because you can't create sounds that weren't on on the CD to begin with. Of course playing back files through an audiophile grade audio system will sound better (with no changes at all) so I suspect that this is why some folks swear that that can hear clear improvements when none can actually be possible. Just play the bits without changing them on the best grade gear you can get and you're all set.
BitPerfect is an outdated player that doesn't seem to perform any function anymore since iTunes doesn't change the bits (sample rate or dynamic range) any more. iTunes did change the bits in the past because we used to go into the Preferences pane and muck around with small rates there. That dates back to the old days where storage systems were tiny and putting an entire library of CDs on your personal computer would fill the drive yup completely. Obviously that is not a problem any more so that reasoning is moot. iTunes now handles FLAC files natively so that reasoning is also moot. In fact if you look in the Preferences pane of any reasonably modern Mac you'll see thither is no longer a setting to change the bit rate of any music when you import it, so whatever bits you get it will be the bits that go out. Save the $10 and out it towards some good music tracks instead.
@E. O. Well FLAC is still not an international standard and it still uses contested IP, so its not standard and can be shut down with a single lawsuit. That is why iTunes has Apple Lossless which is reliable and cannot be shut down.
First time i've watched some of these video's from here in the UK and i have to say that I am very impressed by the methodical way that Paul explains everything. Peoper old school with no bull shite!
Will bewatching more regularly as I'm very passionate about good sound quality. Especially at Home and in my Car.
Hahahaha... you are serious that you care about audio quality in your car?
I care about sound quality in the shower, in underground tunnels and when traveling by train.
I have been ripping CD's using the Mac Finder and getting the AIFF files instead of using an actual ripper from iTunes or some other player. Amongst the agreements and disagreements smoke over AIFF vs FLAC I figured storage is cheap so just go with AIFF. Furthermore, compared to WAV I can store meta data and album art in each track and customize that.
dBpoweramp serves me well.
One of the best questions and answers around. Still, its a standard answer in general though. Increasing any technical levels will not make it better than the original recording.
If you have 5 dollars in your pocket, putting into into a bigger pocket will not make it 10 dollars. Also, It might just get lost.
Storage is cheap, Paul - even for a cheap bastard like me. I use AIFF for archiving but I'll listen to ANYTHING : unlike the snobs, I don't turn up my nose at the 320K Spotify Premium downloads on my laptop. At 60, the worst link in the chain is always going to be my ears - as much as it pains me to admit that - and 30 seconds with a frequency sweep confirms that I just cant hear what I could when I did the initial hearing tests for the Army as a youngster. IMO, the best way to rip CDs is with Exact Audio Copy - ymmv.
Thanks for the video.
I just ripped all my moms cds to 128kbps MP3 files and I'm wondering what the difference is? Playing the song from the cd in windows media player sounds the same to me as playing the song from the rip (in WMP not grove music)
there are differences, yes.
128 or WAV yes, and it depends the speakers or component audio system you have to play the audio.
my sellected playlists with my favourite music commonly is in CDs I record - sometimes the original is mp3, I convert ot to WAVE, I make some improvements if it was necessary and I get my personal CD witth good quality. sometimes the USB with mp3 is enough and I enjoy it too.
@@Ephemeral2023 some songs or videos from youtube (and other different formats I could have) are a mess with poor quality. I have songs since Audiogalaxy or Napster times, even some iTunes that I've purchased need some "restoration": to apply some filters and saving them as WAV files.
@@Ephemeral2023 in 40 years I've listened music, MP3, WAV, AAC, cassette, LP, Minidisc,
I enjoy my music.
My mother and my neighbor enjoy this or those and their music,
but there is a difference between listen in Mono AM RADIO mode than FM Stereo,
_it is better to enjoy more_
commonly it is said as
_it sounds better_
songs I ENJOY sound better, a few times I save the file as MP3.
similar to the movie on TV,
"the important is to enjoy the movie"
yes, but with a better high quality format is better than VHS.
🤷✈🔊
@@Ephemeral2023 no. really I don't.
when I heard the difference between a youtube file and a WAV file after enhance the waveform, and sometimes cleaning up the original file (even from CD tracks), I get a cleaner and better audio file, and I see majoroty of tracks are clipped. of course, I like the process, I have the time and tools to do it with music I'm intereted and I enjoy, because I see the difference between a pig with lipstick and without it. form yuour own point of view to you, it is useless, because you aren't interested, you don't have the ABCD and you don't need it. if you have a different way to listen whatever, I like mine.
Separation. You don't hear separation on a compressed file like on a CD. But you'll need high quality headphones (not earphones) for that.
Though certainly not an expert (I don't even rise to the rank of novice or beginner), I can think of a reason that 24-bit, 96k sample rate could be helpful. Think of it like video frame rates. 24 (film) vs 30 (old video), vs 60 vs 120 frames per second. Suppose the original recording was at 24 FPS for film. If you watch that 24 FPS video frame-by-frame, you see noticeable jump between the frames. While you can't get any NEW information, you can do a linear interpolation between the 2 samples to have a smoother transition between the frames. If you're up-sampling from 24 to 120 FPS, you now have 4 additional frames that have been interpolated between the 2 reference frames. Then, viewing the 120 FPS video frame-by-frame, there's less jump between frames, so it would look smoother. I imagine it could be the same with audio. Finally, with advanced signal processing, not only could you do linear interpolation, but you could also consider the samples before and after the ones between which you're interpolating, and maybe fill in the missing samples based on the preceding and following flow of the music.
Interesting to see the comments that decompressing FLAC only uses a small processing overhead, and that a phone could deal with loads of tracks at once. Also disappointed that Apple will only allow it to run in the OS on iPhone 7 and above as it drains the battery too fast on a weaker chip.
I can play FLACs on my 6s+ with FE File Explorer, I also use it to load my mp3's, FLACs, and MKV/MP4 videos from my NAS to the phone/iPad for travelling
I use flac exclusively
Me too. MP3's are so 90s and the quality really lacks.
@@Snowwie88 I can\t tell a difference between FLAC and 320 kbps MP3. Still, idiot as I am, I stick to FLAC in case I ever buy equipment good enough to actually hear a difference.
Snowwie lacks?
@@B1tterAndThenSome Plus Flac is better as it has no loss either 👍
Hard Drive space is a non issue in 2021. Rip CDs in WAV.
I'm sorry, I was looking for information on ripping CDs, and I seem to have stumbled onto the Alzheimer's Support Network.
In my day we just had wav files. That was it. When flac, aac, ogg, wma, alac, etc.. came out we was like what the hell is this crap?
Back in my day we didn’t even have computers. LMAO
@@speedythecat07 I doubt you are 200 years old. LOL
dBPoweramp is the program to use to rip your CDs or batch convert your digital files of any kind. Exact Audio Copy is OK, but isn't quite as good at giving you exceptional "Bit Perfect" results as consistently as dBPa is able to .
AIFF is Audio Interchange File Format. It’s a pure pcm codec without compression it’s not apple related.
It is Apple related. It was developed by them in the late 1980s.
I used to use video decrypter and encoder suites to copy dvd/on tascam burner for copying cds
What happened to CDA? Isn't the format of the data on an audio CD recorded as CDA as in compact disc audio format? With the size off modern drives being so huge what is the matter with just recording it as a CDA file? That way the bits are exactly the same as on the compact disc, right?
FLAC is more supported for playback. And has better metadata. And each song is a regular file so programs can recognize them better.
this seems like an awesome company. Knowledge is always key.
Did 0:40 make you curious,
So then end up touching with your finger the inner grid even after the button was released and find that it hurts or at least suprisingly gave enough to be uncomfortabke enough to not want to do that ever again if avoidable/if not an inadvertent thing to have happen again at least not on purpose...
*I did*
I normally play sound system on mac at 16/44 no dolby and when I use 24 bit files I use my mackie firewire mixer and audio technica headphones. ... but bitperfect sounds like a neat solution so you don't have to keep switching .
4:20 I would love to hear an elaboration on this concept. Especially, "a revealing system." I find this incredibly interesting. Thanks for the information :)
Do you think that different programs sound different (assuming it's not doing any processing to the file)? I've heard people say that they do but I haven't been able to hear a difference in my own testing and I don't see how it could if the program is just feeding identical data into a buffer for the audio driver. I assume that there isn't any sort of clocking involved at that stage. I can see how using a different driver could make a difference but not a difference playback program.
I'm not talking about resampling or any other processing. Just playing a file bit transparently. Some people say that some players like JRiver still sound better.
*Use Apple iTunes(Music) app on Mac to copy CDs 1:1 lossless in ALAC format.///_* Totally agree with Paul, upsampling is pointless._
You don’t get log with iTunes.
iTunes bad, XLD>>>>
On a computer @ 192kbps if you have a lot of songs I find I get more on my Sony 64gb device at that ripping bit rate, and I recommend using the SonyMediaGo app and with all that Sony give in their sound settings on my device there is not a massive loss in detail by ripping at a lower bit rate👍😃👍 Flac this Flac that blah blah better sound etc etc but like I said before at that vastly high bit rate your MP3 player is going to fill up pretty quickly if you like a lot of songs, so beware go down to 320kbps or do as I do 192 or a happy medium at 256 the choice is yours 👍
just get a bigger sd card for your player
Rip to wav using dBpoweramp. Then do the conversion to Flac after. That way jitter can not enter due to the simultaneous demand of the psu and computer processor. I also close all apps and background tasks. Use the securerip feature (which checks your rip against there database of rips) for errors and if there are any it re-rips multiple passes at lower drive speeds till to improve it. Not add cd's are flawless and it isn't long before you have the odd scratch. Personally what I do if I cannot rip at 99% or greater accuracy I will torrent the audio cd. ( I own the album on at least 2 formats).
I’ve been looking for this answer for years! Thankyou.
Many folks simply use their Windows Media Player to rip and archive CD's to 'lossless' form. They choose 'highest quality' to rip to WMA lossless. I do that as well, especially when I sync my portable FiiO player to my PC, in order to transfer lossless files to it. What are the advantages or disadvantages to simply using the Windows Media Player for ripping CD's in lossless?
Hi Jeremy. Sometime, on some of my PC's, it does! But lately, I have been using my laptop with Win 7, and The sync only takes a few seconds. On my other Win 7 and XP machines, it takes much longer, and sometimes doesn't even sync up at all. I also use an FiiO X1, and love it! That said, I may just try 'Media Monkey'. Thanks.
How does WMP verify the integrity of the CD data? Try ripping the CD several times and I bet you end up with slightly different files.
I'm not sure what you mean exactly by 'integrity of the CD data', but if my ears are any indication, the ripped music sounds true to the original.
If there were a few bits missing in the copy you may not notice it, but it does in fact happen, especially if the disc is damaged or deteriorating. I had several that had to be ripped several times to pass the verification check, and several that were damaged and never did, one of which I had to repair the dropouts.
I mistakenly said EAC before but it's it's actually AccurateRip that Foobar2000 uses now, though I think I had to install EAC to get that feature. See also www.accuraterip.com/
DVD players in computers (maybe not all, I don't know) can't even play CDs slow, so it's even a bigger problem now. I'm pretty sure I switched to AccurateRip/EAC when I eventually noticed a dropout in one of my CD tracks.
Interesting, to say the least. Thanks for the quick education on the subject!
Err..EAC is NOT necessarily the best, but does tend to get the most detail from the disc. Depending on how that info is outputted will determine whether it soundstoo digitally harsh, for me, when streaming 16/44 EAC wavs, it totally did. BUT, for re-burning CD's all that top end information is a hella good, as the process itself will incur some loss just by the burning process being largely mechanical (but huge gains else where). For ripping to stream 16/44, I personally only use XLD on MAC, much more forgiving on the ears, on my system in my room. Try it, they sound waay different.
Wouldn't FLAC be less demanding on the server as well?
Ye, and one thing that wasn't mentioned. Transporting wav files is slower due to the size. So id someone has slow discs, bad wfi etc. It takes a bit more time. So could cause stutters. Also, it is twice as expensive to store!
In other words if the source is analog, sample @ 24-96. But if the source is 16-44.1 digital, just copy in the native format. It's like in photography, they sell us numbers. Who needs a 45X zoom? Not me....
OK - I ripped a CD to iTunes on Mac and set it to AIFF. I now see the files say AIFF-C
What the heck is AIFF-C and how does it compare to AIFF or WAV?
looks like aiff-c used some compression accordig to wiki
What about AFLAC?
So what's the best way to rip a CD to FLAC on a Mac? Freac?
I use music center for PC to rip my CD to flac. It is the best software and most intuitive software I have used.
@E. O. EAC? Is that the new format or the software?
should I keep ripping my CDs to Apple Lossless or switch to AIFF? I recently tested both and can hear a subtle difference, in drums and vocals, but not sure if I'm tripping
First rip and save them in WAV formation (exact original CD quality) now you can rip any formats from that WAV files
rip to uncompressed FLAC
I use AUDACITY for any/all of my CD recordings and then add DOLBY C from my cassettes for that professional sound onto the CD and it sounds great and all you do is take the large end from the cable and plug into the hard,drive and then take the 3mm jack and plug it into the headphones output and then record your CD from the computer.The DOLBY B or C makes the hiss much lower when you play it back.
🤔
Uh, no. Dolby NR is not a single-ended noise reduction process. It is only applicable to recordings that were originally made using the process at the recording end and then during playback. Otherwise, the equalization will be totally whacked. The result would be similar to playing an LP record without RIAA equalization in the phono preamp. BTW, Dolby NR is obsolete and no longer licensed by the company. No contemporary commercial tape recording can be made with the process and therefore should never be played back with Dolby NR. If you have an old tape deck that incorporated Dolby B/C then you could still record with it turned on and then play it back with it also engaged.
Did you forget the green marker pen? ;-)
Have you tried treating the original and blank CD with Auric Illuminator before making a copy?
sample rate matters
CD's are brickwalled at 22.05 kHz; hence the 44.1 sampling rate.
What do you mean by "brickwalled"?
Use AIF most of the times and other files ,Flac ,high resolution some DSD and use ROON which is by far the best software for music and can play any files and is regularly up-grated (best investment I ever did) As external hard disks became reasonable in price I do not see the need to use Flac or ALAC .
All you need is a good burning application, like Burning Studio,to burn files or Rip CDS This is the program I use to burn files or Rip CDs.Even burn movies
personally i use a program called audiograbber and rip the files in .wav . it's an older program but performs in a very efficient manner. it will also compress into .mp3 but i don't use that feature nor the"normalize" function as i prefer to store and playback the way that the engineer and artist created the stored content.
There is such thing as "non destructive replay gain" where the gain values are stored in metadata on a track and album level. This has no effect on the encoded files audio. You can turn of this "replay gain" function within your playback/decoder software.
Audiograbber is all I use myself. I paid for it years before it was turned into a free program and development stopped. It still works great, basic and easy to use. It is really fast on newer computers, compared to the old top of the line 286, 386 and 486 computers when it started.
With my system set for foobar2000 on-the-fly conversion of audio files to 512DSD, I have found that DBpoweramp CD Ripper ripping to flac file setting 'no compression' produces a file which is identical to the sound of the CD used for the rip: any setting other than 'uncompressed' produces a less-resolved sound. I realize that the cost is increased bytes of storage, but my priority is the best possible sound when making or listening to CDs or audio files: drive.google.com/drive/folders/1C1w4HcZuThrTxImadujOdftffK3s_2K3?usp=sharing
JRiver is the best sounding ripping/playback software out there and it's only $60.
I used to chase flies around furiously. And then I thought "All creatures have a right to be" and I ignored them. The odd thing is now they RARELY give me any trouble at all...
huh?
There is no difference between WAV, FLAC, ALAC. They are audibly equivalent storage of uncompressed PCM data. If you hear a difference, it is confirmation bias. Use a tool called Audio Diff Maker, which shows the difference between any two audio files. There may be differences in playback hardware, but not the files themselves. For example, some DAC's clip at 0dBFS, others do not. It is DAC chip dependent. There is also inter-sample clipping. That is, clipping above 0dBFS BETWEEN discrete sample points. Plus, this phenomenon would be exhibited no matter what file format was being played. So, there can be some reasons that HARDWARE sounds different, but the files themselves are identical in every aspect.
You carve the wood by tool or with the machine. The result will be the same at first sight... Proces by handtool is slower more precise and silent, machine is much faster and noisy but not so precise since it is designed to use its power for speed, not precisions and fine carvings. Most of the people will be satisfied with the former but not everyone. So, your software shows only the curve of the sound because software actually cannot show music - only ear can. So I don't know what you and people like you try to prove especially with software. Look at. perhaps, ringing that Bob Stuart claims that affect the sound. Where is that in your software. Pre and post ringing when you turn digital bits into analog electrical wave??? I had discussion recently with the person that brought me into the audio and he told me that crossover doesn't pass the signal higher or lower than cut off freq. at all and I didn't agree and ask him to explain that from technical point of view and what are the first, second and third order crossovers? He had a misconception based upon common snese. My point is that engineers turn theory into product with the topology, chips, capacitors and millions of parts and I'm not nearly capable to put that in question with simple high school physic and "common sense". I can choose to believe or not to believe. Or I can try and decide. Software is redundant. And if you're so sure about your standpoint you should be happy that you're able to save some money without any loss. I suppose you're all altruistic and thank you for that.
Freekwo 777 The software actually does show the "music." Music is only the summed collection of sine waves at different frequencies and phase at the same relative time. There is no difference between the FFT impulse and the deconvoluted frequency response. They are both sides of the same coin. Frequency, Phase, Time, are all the same thing looked from different angles. They are easily represented and these representations are used by engineers every day to build the products you love and admire. As for your friend, he may have been a prime example of someone misguided by wrong info or not understanding what was told to him. I see it everywhere. Good luck in your quest.
Maybe you're not seeing it but you have just made the same mistake as my friend did but on the higher level of knowledge. Music simply cannot be seen, we sense music by ears so your fundamental mistake is more than obvious. When we talking music here we're talking about reproduction and the whole "Hi-fi" means high fidelity of reproduction. Maybe you should go to the concert with you cell phone and record the concert and suggest everyone to the same instead buying pro records on cd, vinyl because software shows the same curve and there is somekind of bias. Maybe Hollywood should do the same with video since some software would certainly show the colour scale the same. Or maybe the old classic movies are not good enough because average cell phone can do the better job nowdays. You're argument has completly miss the point and I'm glad that I stepped in this discussion because there are so many people that confusing other people with self made philosophy without any praxis and now we know that software shows music. Thank you for that fact. fortunately it's only a link in the audio chain and certainly isn't music since I never heard the curve.
Freekwo 777 Ok. Now I am getting to the point where it is hard to understand you. If music cannot be seen, measured, or quantified, then all the audio designers, engineers and sound techs, mixers, masters, and performers should just throw away all test gear and design by ear? Your statement bears nothing in fact, only subjectivism. History has taught us that subjectivism, "feels" if you will, is always wrong. Certainly music has an emotional element as we connect to music emotionally. However, we can certainly see and calculate waveforms, convolve and deconvolve impulse responses and SEE the outcome. By comparison, we can also SEE the effects of change. Why do you suppose EVERY designer of audio gear uses TOOLS, save for the layperson designing crap in their garage with a multimeter and soundcard "analyzer"? Dispell with mythology my friend.
"We should no more let numbers define audio quality than we should let chemical analysis be the arbiter of fine wines." Nelson Pass
The hell is this....no, Paul, decoding ALAC files will not "jitter the power supply" to impact sound quality. This is just dishonest information.
And he's wrong about WAV metadata.
As a derivative of RIFF, WAV files can be tagged with metadata in the INFO chunk. In addition, WAV files can embed any kind of metadata, including but not limited to Extensible Metadata Platform (XMP) data or ID3 tags in extra chunks. Applications may not handle this extra information or may expect to see it in a particular place. Although the RIFF specification requires that applications ignore chunks they do not recognize, some applications are confused by additional chunks.
Video's junk.
What was that again with adding random bits for dithering? And that Nyquist hesitations? Nevermind that "flac could sound different than wav". Everyone can have a bad day. I'd just not post on one of those days ;)
FLAC is still the best i recon. Works kn everything, lossless and smaller the WAV.
The F is very important, it's a FREE and open standard
But of course. No problems with it whatsoever.
Not on everything as it wouldn't on my 4th generation iPod touch.
@Mike P if it really is unaltered. I.e. qobuz vs Tidal is interresting. Both say lossless. Qobuz is true lossless and Tidal mqa has been analysed a lot and they add some loudness effects and certainly not lossless. I don't know how apple will deliver yet.
I am on qobuz.
@Mike P good point. I did made an offline copy once (no cost), i ll need to look at where those files are and if they are copy able.
I like to rip CDs onto FLAC (with dbpoweramp) or ALAC (with iTunes) because they support metadata very well. Of course with 16/44.1 as CD standard does, I don't like something upsampled. If I want to hear the 24-bit music I just open my Qobuz or Tidal app
Dbpoweramp is the best.
hey im wanting to rip some sacds to play on sacd player i been told to use normal dvd -R discs any thoughts.
Now if you have a vinyl and you want to put the vinyl on a hard drive, what would you sample it bit at 24bit or 16bit?
I would say most records would be fine sampled at 16 bits. There are some records that actually have a wider dynamic range than will fit within the old CD digital standard however. yes the noise floor on the record can be easily heard during the quiet passages, but so can the music. Recording to 16-bit, setting the peak of the recording to match to the maximum level of the digital standard, the quiet passeges drop to dugital silence with an periodic soybd here and there as the music and surface noise raises up to the lowest level allowed with 16-bits.
I do not know what the dynamic range is with 24-bits...
Another thing to consider is if you plan to do any processing to the audio after recording a record to digital, is that processing can negatively alter the audio very quickly with the low resolution of 16-bit. 24-bit allows the audio to be processed with little errors created while processing with digital filters, equalizers, noise processors, or even simple digital volume adjustments.
One possible advantage of ripping vinyl with a higher bit rate would be the increased headroom before aliasing. I've transcribed over 600 LP's to digital and use the standard 44.1 KHz/16-bit CD format. The main issue I had to deal with is that it can be very time consuming to preview every second of an LP to determine the maximum audio amplitude and set the digital recording device to the highest possible level (for the best resolution) without clipping (anything over 0 dB), which ruins a digital recording. My solution was to try to quickly find the loudest passages and set the recording level appropriately (for maximum resolution) but use a soft-knee limiter with look-ahead delay just ahead of the A/D conversion to knock down any unexpected peaks that would cause clipping. I use a tc electronic Finalizer Express (A/D/A converter) which has the limiter on the analog input. Absent a limiting function on your A/D converter you could just buy an analog limiter and place it ahead of the converter and then digitally record at the sampling rate you use for playback.
I have ripped my cd collection in wav by pc equipment and stored in my home cloud. Knowing that the rip quality cannot be "perfect" due to poor jitter and other factors (electrical/electronic noises) generated in the pc system during the process of ripping. Anyway, just enjoy the cloud music collection streamed by my two oppo players and Chromecast audio. I find wav format is "more musical", just personal opinion.
Use flac
YHO Vo is Apple lossless the same as flac?
@@GabrielMartinez-pe6ln Yep its lossless
you sound schizophrenic
Does this mean that early PC's in the 1988 could not play CD's ?
I used Exact Audio Copy to rip and FLAC. Works for me.
I'd still use 128kbps MP3 and play TH-cam through any hifi speaker I have though :)
Rennie Ash what’s the compression
Quality through songs on TH-cam?
Wait is that fly swatter audiophile quality like the audiophile mask? Huh Paul. Oh cmon Paul
What's the best DA converter to use on my turntable to get the least loss of information?
Brad Tomlin uu
bit perfect the best $10 I've ever spent even with my less than audiophile basic system.
I use iTunes. Don't crucify me