Could you help me setup how to rip CDs into FLACs?

[Chrispy]

This theory is similar to one applied in digital photography. Photograph your media in a slightly overexposed/blown out shot, then in processing, bring the entire level down to "being what you wanted". The idea is that on a "pixel/data" scale, you will have more data available to adjust if shot brighter, vs the data available in a darker pixel. I can understand how an audio file can be made better with the same principle.



Thanks for that. Thats some pretty good tips for problem childs. I'm about to sit down and spend some time looking at dbpoweramp, then I feel like starting some past time rips today.



Thanks for this comment. I think you kind of brought a subject to light for me, that I didint realize about EACs use.

I remember when I first started using EAC I set error recovery to high, then changed to low and rip times improved. Haven't thought a lot about it since. Altho my initial rip of my collection (done while couch bound with a knee injury) did include some damaged discs, but think I did most of those in the beginning to see if they ripped okay. These days I just on receipt of a new disc play it once, rip it and file it, don't even pay attention to how long it takes.

 

[PjV]

I do my ripping on a Mac using XLD. It rips using the AccurateRip database which means it will rip once if it can verify the rip is good using the db. Otherwise it rips twice. I only rip at 1x if there’s some problem reading the CD otherwise it’s full speed which ends up being like 8-14x.

Then use Picard to add metadata.

I’m sure there’s a similar setup on PC.

I use the same combination. On PC I used to use EAC.
Next to the Apple drive is a noname dvd drive, for the rare -copy protected- CD that doesn't rip without errors.
Usually re-ripping the faulty tracks on the other drive works well.

For backup I synchronize the files (via OneDrive) to a nas on a different location.

 

[Massiveattack]

So I did decide to buy DBpoweramp and I have to say, I already like it. It somewhat feels familiar like EAC menus a bit, its MUCH easier to select how you want to rip your CD, and generally it just feels so much more user friendly.

You dont have many options for FLAC, obviously, but I noticed that there is the ability to Encode or "compress" at varying levels, to be able to save some disk space.

I ripped a test CD (Massive Attack - Mezzanine, what else) and did a full level 8 encoded FLAC in about 2mins 20 second! At this rate ill have my entire collection re-ripped in a Saturday!

Uncompressed was 675mb
level 8 compressed was 375mb
CBR 320kb mp3 was 147mb
CBR 192kb mp3 was 89mb

I really like EAC, but I am clearly not smart enough to use it, I need something dumbed down. It looks like DBPA will be a nice addition. Time to pull out all my CDs!



Thank you everyone for this little guidance you all gave me. I seem to have gotten my answer in a different form, it was easier to just go ahead and move to another program.

 

[Miguelón]

Just for your possible reference and interests, I assume;
- Summary of rationales for "on-the-fly (real-time)" conversion of all music tracks (including 1 bit DSD tracks) into 88.2 kHz or 96 kHz PCM format for DSP (XO/EQ) processing: #532

Spoiler

in post #532 on my project thread, I wrote as follows...

Summary of rationales for "on-the-fly (real-time)" conversion of all music tracks (including 1 bit DSD tracks) into 88.2 kHz or 96 kHz PCM format for DSP (XO/EQ) processing

Hello @adLuke san,

Welcome to this project thread! Your above inquiry is nice and important point, indeed.

My present answer for you is "It is quite feasible enough and even ""needed"" to feed all the audio digital signals in 88.2 kHz or 96 kHz PCM (or 192 kHz, if you like) by JRiver's on-the-fly format conversion to be sent into DSP (XO/EQ) software EKIO. "

Various background and justifications for this answer are as follows;

Before starting this project, I had been enjoying music with ordinary PC audio setup with one DAC (OPPO Sonica DAC)) and one HiFi integrated amplifier (ACCUPHASE E-460) driving all the SPs through passive LC (inductors capacitors resistors) network. And I had been sticking to "native format feed" into OPPO Sonica DAC up to 1-bit/DSD256(4x), as you kindly pointed.

When I started considering possible multichannel multi-driver multi-way multi-amplifier project with software DSP (XO/EQ), I did intensive search and desk evaluations on various DSP software solutions, and I found the maximum PCM processing format is 192 kHz 24 bit in these DSP software solutions. (Even with the extraordinary expensive TRINNOV ALTITUDE 32 DSP processor, actually having PC in it, the internal DSP processing is up to 192 kHz).

I carefully considered the pros and cons of "DSP processing all tracks in 192 kHz or 96kHz" instead of "native format feed", and concluded that multichannel multi-amplifier approach would surpass the cons, at least in my system setup with still amazingly wonderful Yamaha SP drivers and cabinet.

Consequently, I decided to go into "multichannel multi-amplifier" world of "max. 192 kHz 24 bit processing", as you kindly have read through this project, including the "all in max. 192 kHz ASIO I/O within PC".

Then, rather recently, I (we) fully discussed and evaluated the UHF (ultra-high frequency) noise issue in poorly QC-ed HiRes music tracks including DSD formats, as you clearly noticed;
- "Near ultrasound - ultrasound" ultra-high frequency (UHF) noises in improperly engineered/processed HiRes music tracks, and EKIO's XO-EQ configuration to cut-off such noises: #362-#386, #518
I wrote that such a high amount of UHF noises would be "possibly" harmful (and useless, meaningless) for our tweeters and super tweeters. I also pointed they would be highly possibly harmful for our beloved pets including dogs, cats, birds.

Having my intensive objective measurements of these "poorly QC-ed" HiRes tracks, and having so many intensive discussions on "enough PCM sampling rate in HiFi audio", now I conclude that 88.2 kHz or 96 kHz processing (i.e. up to 44.1 kHz or 48 kHz in L and R channels) would be just enough and feasible in my setup (and I believe so also in your setup) since I decided always having high-cut (low-pass) -48 dB/Oct filters at 25 kHz in my EKIO configuration to cut-off any of the possible UHF noises very frequently existing in HiRes tracks.

This means that I have finally landed on agreement with @mikessi's "enlightenment and belief" of "There is really no audible benefit to playback beyond 24/96 sampling, especially with any recordings other that those done with the most advanced high res gear and high fidelity values."　

Another important aspect of this issue would be relating to our hearing ability in high frequency zones. Recently, I participated in the interesting thread entitled "Audio Listening With Age Diminished Hearing". You would please read my posts #70, #72 and #74 on that thread.

BTW, as I wrote here, here and here, my digital music library of about 25,000 files consists of mixture of various formats;

16-bit/44.1kHz CD ripped non-compressed aif (majority!),
24-bit/192kHz down-sampled or up-sampled aif,
24-bit/96kHz flac,
24-bit/192kHz flac,
1-bit/DSD64(1x) 2.8MHz dsf,
1-bit/DSD128(2x) 5.6 MHz dsf,
1-bit/DSD256(4x) 11.2 MHz dsf,

and now JRiver MC feeds all of the tracks usually (mainly) in 88.2 kHz 24 bit (i.e. max. 44.1 kHz Fq window in 2-ch stereo) by on-the-fly conversion into EKIO for crossover/EQ processing. As I have high-cut (low-pass) -48 dB/Oct LR filters at 25 kHz, max. 44.1 kHz in L & R channels are more than enough.

Yes, I agree about upsampling if one considers DSP treatment as possible cause of clipping and reducing dynamic range.

But if OP wants to shorten the rip time per album, it will last even more on conversion, am I wrong?

 

[Miguelón]

Uncompressed was 675mb
level 8 compressed was 375mb

The level of compression implies lose of fidelity?

 

[recycle]

24/96 is not better quality than 16/44.1: just provides more unnecessary information. Rebuy 200 CDs will be expensive and on 24/96 it occupies same space as 600 CD with no additional quality (as much as 420 Gb).

For someone whose nickname is “recycle”, a full contradiction

Yes, 96/24 is definitely better than 44/16

 

[Miguelón]

Yes, 96/24 is definitely better than 44/16

I wont go on the eternal absurd discussion about human capacity to distinguish between 16/48 and 24/96. It has been finished and closed.

Only on recording and studio mastering will be useful, you cannot distinguish between Hi-Res and CD. If you think you can, make a proper blind test

 

[Graham849]

I use Windows 11 media player (legacy) to rip cd's to FLAC, much faster than EAC which I tried three years ago. At my age 16/44.1 makes me content. ymmv

/enjoy the music

 

[Graham849]

Yes, 96/24 is definitely better than 44/16

from a CD, you jest m8

 

[Vincent Kars]

The level of compression implies lose of fidelity?

No.
The word compression causes a lot of confusion as it is associated with lossy compression.
The compression parameter simply tells how many CPU FLAC is allowed to use to find the best possible linear prediction. The better the prediction, the lower the residue will be hence a smaller file.

When Josh Coalson started the development of FLAC in 2000 this parameter made sense. Lots of PC's not even strong enough to run XP.
With today’s CPU’s you probably won’t notice the difference in time between e.g. 0 and 8.
The I/O is likely the limiting factor, not the CPU.

Regardless of the compression ratio chosen, the result is always lossless.

Bit more detail

 

[Miguelón]

No.
The word compression causes a lot of confusion as it is associated with lossy compression.
The compression parameter simply tells how many CPU FLAC is allowed to use to find the best possible linear prediction. The better the prediction, the lower the residue will be hence a smaller file.

When Josh Coalson started the development of FLAC in 2000 this parameter made sense. Lots of PC's not even strong enough to run XP.
With today’s CPU’s you probably won’t notice the difference in time between e.g. 0 and 8.
The I/O is likely the limiting factor, not the CPU.

Regardless of the compression ratio chosen, the result is always lossless.

Bit more detail

Perfect, thanks for clarifying.

 

[recycle]

I wont go on the eternal absurd discussion about human capacity to distinguish between 16/48 and 24/96. It has been finished and closed.

Only on recording and studio mastering will be useful, you cannot distinguish between Hi-Res and CD. If you think you can, make a proper blind test

When you say "Only on recording and studio mastering will be useful" you are implicitly confirming that 96khz/24bit sounds better than 44khz/16bit.
In fact it is exactly like that

 

[dwkdnvr]

I was facing the question of what to do with my CD collection. I'd already ripped some of it manually ('the good stuff'), but only maybe 30-40%. It was either take the time to rip, or dump them and just rely on streaming.
I set up 3 Linux PCs with 'Automated Ripping Machine' installations and 2 drives each. Worked amazingly well. Ripped 500+ CDs in very short order - just wander back every 20 minutes and swap in a new batch of 6 disks.
You do have to set up the metadata config in ARM to ensure you capture the important details, but I access my ripped collection via Roon, which might make the metadata question a little less critical.

 

[Miguelón]

When you say "Only on recording and studio mastering will be useful" you are implicitly confirming that 96khz/24bit sounds better than 44khz/16bit.
In fact it is exactly like that

I don’t, is a question of recording and manipulating the file, and preserving head room. Normally digital files are recorded on microphones and transmitted to an ADC which converts them to digital numbers.

Is a little bit long to explain you the process, but to avoid microphones or ADC clipping, the alignment level should be placed below the maximum, taking place of the remaining bits and therefore diminishing dynamic range.

When over sampling to 24 bits, you increase the dynamic range and low the noise level, and at the same time leaving head room for the system ti work.

With sample rate is different: on mixers and other editing process you will do some mathematical treatment that results in lower resolution final file (mainly Fourier transformations and reverses if you want to learn). Upsampling to 96 kHz let this degradation on the Hi-Res file acceptable (non audible) on CD format after down sampling.

This are technical treatment that don’t produce a home setup, and the technicians are not capable neither to distinguish the resolution. Their audio interfaces, mixing consoles and stuff will be programmed to work at this level, that’s all

 

[recycle]

I don’t, is a question of recording and manipulating the file, and preserving head room. Normally digital files are recorded on microphones and transmitted to an ADC which converts them to digital numbers.

Is a little bit long to explain you the process, but to avoid microphones or ADC clipping, the alignment level should be placed below the maximum, taking place of the remaining bits and therefore diminishing dynamic range.

When over sampling to 24 bits, you increase the dynamic range and low the noise level, and at the same time leaving head room for the system ti work.

With sample rate is different: on mixers and other editing process you will do some mathematical treatment that results in lower resolution final file (mainly Fourier transformations and reverses if you want to learn). Upsampling to 96 kHz let this degradation on the Hi-Res file acceptable (non audible) on CD format after down sampling.

This are technical treatment that don’t produce a home setup, and the technicians are not capable neither to distinguish the resolution. Their audio interfaces, mixing consoles and stuff will be programmed to work at this level, that’s all

You assume that the file will be used on a home setup, which is not always true: there are people who play Flac files for professional pourposes and therefore need the highest resolution possible to avoid degrading the sound in subsequent processes: I use Flac exactly for this reason.
when instead I have to listen to an album in my portable boombox during the barbecue then the high resolution file has no meaning anymore.
In any case, I prefer to have Flac Hi Res no matter what use I will make of it, this to have a future-proof support

 

[Miguelón]

You assume that the file will be used on a home setup, which is not always true: there are people who play Flac files for professional pourposes and therefore need the highest resolution possible to avoid degrading the sound in subsequent processes: I use Flac exactly for this reason.
when instead I have to listen to an album in my portable boombox during the barbecue then the high resolution file has no meaning anymore.
In any case, I prefer to have Flac Hi Res no matter what use I will make of it, this to have a future-proof support

Is true in the case of the OP: he will demand advises to rip his CDs to FLAC to use them on a DAP.

In which cases you need FLAC on 24/96 for playing (weather professional or not)?

Even at most exigent and aggressive multichannel equalization you can upsample your master with software. I think there exists DACs that systematically transforms each file to 24/192 kHz, if I’m not wrong even WiiM Ultra do that to preserve dynamic range and headroom when EQ or changing digital volume.

No need to rip on any other resolution than CD, is a nonsense

 

[Vincent Kars]

I think there exists DACs that systematically transforms each file to 24/192

Except stated as explicitly NOS (Non Over Sampling) almost all DAC's do up- or over-sampling.

Basically the bit depth is about the arithmetic precision of the data path between source and DAC. If it is 16, you do have to dither (adding noise) to get rid of the quantization error inherent to any kind of DSP including digital volume control. If you switch to 24, the LSB is at -144 dBFS. As there is no playback chain able to resolve this ( modern power amps can resolve 20 bits max), you don't have the quantization error problem.

Over-sampling is stone age. Already in the early 90's CD players used 4 and later on 8 times oversampling. This simply pushes the alias ( 4x 44.1=176.4) way out the audible range and very likely out of the range your amps can reproduce.

 

[Miguelón]

Except stated as explicitly NOS (Non Over Sampling) almost all DAC's do up- or over-sampling.

Basically the bit depth is about the arithmetic precision of the data path between source and DAC. If it is 16, you do have to dither (adding noise) to get rid of the quantization error inherent to any kind of DSP including digital volume control. If you switch to 24, the LSB is at -144 dBFS. As there is no playback chain able to resolve this ( modern power amps can resolve 20 bits max), you don't have the quantization error problem.

Over-sampling is stone age. Already in the early 90's CD players used 4 and later on 8 times oversampling. This simply pushes the alias ( 4x 44.1=176.4) way out the audible range and very likely out of the range your amps can reproduce.

Thanks for explanation, I could imagine by myself because reducing volume itself will reduce dynamic range on 16 bits dramatically. I’m on learning process of sound treatment, not a lot of security on knowledge but begin to orient myself
Is about 1 bit per 6 dBFS, I’m right? So reducing 24 dB lets 12 bits to encode signal strength and augments quantization noise floor by the same 24 dBs…?

 

[bumblebee657]

When you say "Only on recording and studio mastering will be useful" you are implicitly confirming that 96khz/24bit sounds better than 44khz/16bit.
In fact it is exactly like that

Be aware that the HiRes versions of old albums are often from the worst remaster available. Lots of them sound horrible without any dynamics. Just check out the "DR database" and you'll see that the HiRes versions are usually among the worst. I usually look for the first release when I get (used) CDs. Not all remasters are bad, but I'm alarmed when I read "remaster".

 

[pseudoid]

I elected myself to remind you of 2 matters that are NOT trivial:
#1 >> Back-up (and/or archive) right-upon transcoding.*
#2 >> Start by creating your own DeweyDecimal system for labeling each CD's output, and each folder/subfolder you will put them in.
*extra-point for starting such a system in NAS, with at least 10X capacity as you currently think you need.