Do files ripped from CD sound differet if they are FLAC vs MP3

[yepimonfire]

But I thought this is just a visual representation, a graph. I didn't think anything can be literally square shaped in sound reproduction.



Don't think you'll scare me away with this sentence! These quotation marks worry me. The term theoretical is used to describe just about everything from: "not true at all and I'm just throwing it out there" to "verifiable by hard data". I was hoping for the latter, but not all members seem to agree with you.

I'm interested in relations between digital domain and audible sound and haven't reached that chapter in "Sound Reproduction".

What makes it slightly more difficult is the fact that some terms are used in both, like frequency. As I said, someone skipped directly from frequency of sampling rates to what is audible. Which led me to believe that there's certain causality among them.

Hence:

Do you need high sampling rates to get a very high pitch/FREQ (not only what is enough, but ever)?

Do you need high sampling rates for music that goes from very quiet to very loud? (Think more than Private Investigations)?

Do you need high sampling rates for the music that goes from very low FREQ to very high FREQ?

And same goes for bits. I read more than once that since vinyl is equivalent to 8bits (10 the most whatever that means), it can never achieve the same dynamics as RBCD. This, at least to me, seems to imply that you need a certain amount of bits to achieve a certain amount of dynamics. So, again, is there a direct correlation?

Do dynamics always imply both huge oscillations in dB and FREQ or is one of these enough for certain material to be considered to have great dynamics?

And what if it's just timbre? (A lot of different instruments playing the same or similar tune, but same loudness and octave)

Is RBCD sometimes big enough to capture everything and sometimes not? Would you ever, ever need a hi-res audio for a Punk band?

No. In digital audio, sounds below the quantization noise floor are lost, however, it’s highly unlikely real music would have a dynamic range large enough to even have sounds lost to the noise floor. Take classical music for example, which has a high dynamic range. An average recording may be centered at -30dBfs. Let’s assume the music has a dynamic range of +-30dB. Peaks may approach 0dBfs, and very soft passages will be down -60dB. If you turned the volume control up high enough to recreate the full spl of a classical music concert (about 95dB on peaks), the softest sounds would play at 35dB, the loudest would play at 95dB, and the average medium sounds would play at 65dB. The average noise floor of a quiet house is between 30dB and 40dBA, anything significantly quieter than this noise floor (for example, 10dB) will be inaudible due to masking, anything significantly quieter than the loudest passage (for example, a -60dB sound buried beneath a -15dB sound) will also be inaudible due to masking. 16bit audio still captures and reproduces these sounds. This is the kind of stuff lossy compression (like mp3) throws away, because you can’t even hear it.

Lots of rock music (including punk) is lucky to have a dynamic range of 12dB, usually less. Most modern recording are compressed within a 12-6dB range, you could probably get away with using 8 bit due to this.

The dynamic range of 16bit is about 96dB, 120dB with dither and noise shaping. There is absolutely no reason you would need a higher dynamic range for playback.

Transients can be smeared at a lower sampling rate due to something called ringing, where the sound does not start and stop instantly. The higher the sampling rate, the shorter the impulse response. This is the whole idea behind mqa, RBCD cannot accurately reproduce transients in the time domain, whether or not this is audible is a subject of hot debate. For me, the fact that I could differentiate a 192khz file between a 44.1khz file during an abx test is enough for me to conclude that it is audible.


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[Irvrobinson]

Do you understand how sampling and quantization works? I suggest you read the articles listed below before resulting to childish insults.

https://en.m.wikipedia.org/wiki/Digital-to-analog_converter

https://en.m.wikipedia.org/wiki/Pulse-code_modulation

http://www.yamahaproaudio.com/global/en/training_support/selftraining/audio_quality/chapter5/09_temporal_resolution/

http://www.yamahaproaudio.com/global/en/training_support/selftraining/audio_quality/chapter5/03_frequency_range/


http://emerald.tufts.edu/programs/mma/mrap/Batesweek6.doc

These are not technical articles. The one from Tufts University is so silly it has grammar errors. (What is an "infinitive" filter? One that filters out verbs?) Just because something is posted on a web page does not mean it is correct.

I now see where you got that silly interpretation of aliasing where samples are "folded down".

You are an output-only device, you are not listening to anything anyone is saying here, and you post non-sense masquerading as technical interpretation. I'm not a moderator, so I can't ban you from posting this silliness, but I can ignore you completely, which starts now.

 

[Alex2507]

Are you stoned?

Are you holding?

 

[yepimonfire]

These are not technical articles. The one from Tufts University is so silly it has grammar errors. (What is an "infinitive" filter? One that filters out verbs?) Just because something is posted on a web page does not mean it is correct.

I now see where you got that silly interpretation of aliasing where samples are "folded down".

You are an output-only device, you are not listening to anything anyone is saying here, and you post non-sense masquerading as technical interpretation. I'm not a moderator, so I can't ban you from posting this silliness, but I can ignore you completely, which starts now.

I actually never said anything about aliasing. I am disagreeing with you because you stated that a transform filter doesn’t smooth the digital waveform, that it comes out of the sac smooth and all it does is remove aliasing, which is entirely incorrect. Aliasing is removed, but the original approximation of the waveform cannot be recovered without the transform filter (low pass), as it contains harmonics that are not a part of the original signal.

See http://instrumentationlab.berkeley.edu/Lab10

A low pass filter does indeed affect impulse response see
http://lavryengineering.com/pdfs/lavry-sampling-theory.pdf

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[sterling shoote]

No. In digital audio, sounds below the quantization noise floor are lost, however, it’s highly unlikely real music would have a dynamic range large enough to even have sounds lost to the noise floor. Take classical music for example, which has a high dynamic range. An average recording may be centered at -30dBfs. Let’s assume the music has a dynamic range of +-30dB. Peaks may approach 0dBfs, and very soft passages will be down -60dB. If you turned the volume control up high enough to recreate the full spl of a classical music concert (about 95dB on peaks), the softest sounds would play at 35dB, the loudest would play at 95dB, and the average medium sounds would play at 65dB. The average noise floor of a quiet house is between 30dB and 40dBA, anything significantly quieter than this noise floor (for example, 10dB) will be inaudible due to masking, anything significantly quieter than the loudest passage (for example, a -60dB sound buried beneath a -15dB sound) will also be inaudible due to masking. 16bit audio still captures and reproduces these sounds. This is the kind of stuff lossy compression (like mp3) throws away, because you can’t even hear it.

Lots of rock music (including punk) is lucky to have a dynamic range of 12dB, usually less. Most modern recording are compressed within a 12-6dB range, you could probably get away with using 8 bit due to this.

The dynamic range of 16bit is about 96dB, 120dB with dither and noise shaping. There is absolutely no reason you would need a higher dynamic range for playback.

Transients can be smeared at a lower sampling rate due to something called ringing, where the sound does not start and stop instantly. The higher the sampling rate, the shorter the impulse response. This is the whole idea behind mqa, RBCD cannot accurately reproduce transients in the time domain, whether or not this is audible is a subject of hot debate. For me, the fact that I could differentiate a 192khz file between a 44.1khz file during an abx test is enough for me to conclude that it is audible.


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Congratulations, sincerely. You are the only audiophile that I have ever heard to say "I could differentiate a 192khz file between a 44.1khz file". I do believe you. Not that it matters. But, can you give details about your ABX test. I'd like to reproduce it for myself.

 

[killdozzer]

Folks, I never meant to stir you up. I'm sorry if I did. I'd rather rely on what I read in the books from now on. Having one person appear totally confident while others oppose significantly is not reassuring to me. Preferences are subjective, but I'd hope that turning digital data into sound would be less so.

I still thank you for your time and effort.

 

[Irvrobinson]

Are you holding?

Absolutely. I'm holding my tongue in the face of abject stupidity.

 

[Steve81]

A low pass filter does indeed affect impulse response see
http://lavryengineering.com/pdfs/lavry-sampling-theory.pdf

I'm curious if you actually read the whole article or not. It's an odd choice to support your position given that the author is specifically making the case against 192kHz, concluding:

Sampling audio signals at 192KHz is about 3 times faster than the optimal rate. It compromises the accuracy which ends up as audio distortions. While there is no up side to operation at excessive speeds, there are further disadvantages:
1. The increased speed causes larger amount of data (impacting data storage and data transmission speed requirements).
2. Operating at 192KHz causes a very significant increase in the required processing power, resulting in very costly gear and/or further compromise in audio quality.

The optimal sample rate should be largely based on the required signal bandwidth. Audio industry salesman have been promoting faster than optimal rates. The promotion of such ideas is based on the fallacy that faster rates yield more accuracy and/or more detail. Weather motivated by profit or ignorance, the promoters, leading the industry in the wrong direction, are stating the opposite of what is true.

There is no mention that the 44.1kHz sample rate of CDs being tremendously inferior as a consumer format, and really the paper seems geared towards how you get there properly as opposed to recommending that we ditch 44.1kHz in the consumer space and make the bump up to a ~60kHz sample rate.

The link doesn't devote much discussion to the use of FIR filters (saved for other literature), except to state they yield an "excellent approximation to a theoretical sinc function" i.e. a brick wall low pass, and a note about a 60kHz sample rate totally eliminating any argument regarding associated pre-ringing as an issue (though the ringing is only around the cutoff frequency and fairly low in level, so it's not a practical issue in any case).

And one last goodie on the topic of microsecond impulses...

So if going as fast as say 88.2 or 96KHz is already faster than the optimal rate, how can we explain the need for 192KHz sampling? Some tried to present it as a benefit due to narrower impulse response: implying either "better ability to locate a sonic impulse in space" or "a more analog like behavior". Such claims show a complete lack of understanding of signal theory fundamentals. We talk about bandwidth when addressing frequency content. We talk about impulse response when dealing with the time domain. Yet they are one of the same. An argument in favor of microsecond impulse is an argument for a Mega Hertz audio system. There is no need for such a system.

Thanks for the link

 

[mtrycrafts]

.... I'd like to reproduce it for myself.

But you may not have his acuity.

 

[sterling shoote]

Folks, I never meant to stir you up. I'm sorry if I did. I'd rather rely on what I read in the books from now on. Having one person appear totally confident while others oppose significantly is not reassuring to me. Preferences are subjective, but I'd hope that turning digital data into sound would be less so.

I still thank you for your time and effort.

Here's what seems to be the consensus, while hi-res can be seen to be better, it cannot be heard to be better. That's to say, most folks don't have the ability to hear any difference between CD and hi-res, while visual audio measuring devices might show hi-res to be better. BTW, there are plenty of folks, like me, who believe that CD is still the reigning champion of audio storage media, and SACD the champ for multi-channel, although there is no promotion for these formats today.

 

[Yohansen]

Some misinformation here, some truth.

CD audio is not compressed, in the sense that information is compromised or lost. It may be dynamically compressed, but that's not the same thing. The CD audio is lossless and is for all practical purposes flawless, in that it perfectly conveys the intent of the artist and the recording engineer without technical compromises that would affect your perception of the audio. It does not degrade over time, and the CD audio technical spec hasn't changed since its inception more than 30 years ago. The CD technical spec is perfectly adequate to convey any waveform perceptible to humans at a more than adequate dynamic range. Mastering techniques have improved over time, but those change what's being encoded, not the adequacy of the encoding method. The CD audio spec is good enough for human hearing, and always will be. Mp3 is a lossy format, and some information is always lost, regardless of the bit rate. However, the difference between a high quality (320 kbps) mp3 and the original audio file is subtle, and may not be perceivable at all depending on the source.

The CD audio is what it is: a perfect representation of the artist's intent, as good as he or she could make it, from his or her brain to your ears. A FLAC is a perfect copy of the CD audio, like a zip file. I like them because they save some space, but mostly because they have better metadata options than WAV files and I can always opt to extract the original WAV file from them if I ever wanted to. A high bit rate mp3 is usually all but indistinguishable from the original, is much smaller than FLAC, has great metadata options, and can be played by anything.

Well said, I agree. I have a 4 TB Hard drive full of flac songs...some or few of them are mp3.

 

[Bucknekked]

Folks, I never meant to stir you up. I'm sorry if I did. I'd rather rely on what I read in the books from now on. Having one person appear totally confident while others oppose significantly is not reassuring to me. Preferences are subjective, but I'd hope that turning digital data into sound would be less so.

I still thank you for your time and effort.

@killdozzer
I think you sell yourself short. Don't give up simply because there are differences of opinion. I think in most hobbies where there are subjective opinions and ideas, folks will differ and be more than willing to support their viewpoint.

The problem with sticking to printed books is that so many of them are filled with hogwash, or, simply theoretical concepts. When you get in to theoreticals about sound the first casualty is often "is it audible?" Many if not most theoretical differences in sound are simply not audible. Even if you look at the differences in specs printed on the spec sheet of an amplifier, many of those differences are not audible either.

When it comes to HD audio and the CD standard, this is the center point of the most of the discussions. There are devotees that firmly believe the HD audio (and all its variations) must be better because they have more bits and information. There are those that wish to denigrate the CD standard because its simply 30 years old and certainly something must be better.

But when it comes to "is there an audible difference", the question does have an answer. You can get that answer for yourself without any "experts" (whether in books or on forums) by simply doing your own A/B tests with your own equipment in your own listening area. Don't cheat yourself and do it one time. Do it a dozen times over a period of days or hours. Those first couple iterations can be filled with expectation bias. After you do it enough times however and you get dialed in to what you are listening to, then the answer will become clearer to you.

I can't tell you what your answer will be. I can tell you what my answer was, but, then again you didn't ask and its not really relevant to getting your own answer. Get your own answer to the question and then you should be comfortable no matter what the experts say.

 

[Yohansen]

That is the best advice I could get.
Thank you

 

[BMXTRIX]

@Bucknekked is on point and is preaching the same thing that I do over and over.

There is a vast difference between the near ridiculous conversation/argument going on in the background to the simple reality that sits before you.

Can YOU hear a difference between a CD (or FLAC) and a MP3 at 320, 256, 128, etc. Kb/s encodes? Or, more accurately, at what compression level do you hear a difference? You may want to try a few different encoders because (apparently) they aren't all built the same.

Listen on your gear. Listen on the best sounding audio platform you can listen on, and switch it up some if you can.

Most of all, at what point do you even care? Even if a 128Kb/s mp3 doesn't sound as good as the CD original, is it a song you care enough about to redo the encoding at a higher bit rate? If you do 90% of your listening in the car, then does the higher bit rate make any difference at all with your typical listening?

The takeaway from all of this is that some people are passionate about the minutiae, others are less so. You determine your own path in all of this, and your own enjoyment.

My opinion is that while I want a decent copy of my music, I'm perfectly happy with 192Kbs on my CDs, but I've upped it to at least 256Kbs or higher because disc space has gotten ridiculously inexpensive and CDs are insanely small these days by comparison. At 600MB per disc, a 6TB hard drive will store 10,000 CDs without any compression to the audio at all. But, adding compression can save you 70% or more with MP3 and a good deal with FLAC (ALAC). So, if you want to put it all on your 128GB phone, then you have to consider that in the process.

Back in the day (80s), I didn't own a CD player, but my father did. I did have a tape deck. So, I bought everything on CD and I copied all of my CDs to cassette tapes. I could listen to them anywhere. It was perfectly fine for casual listening, but when I got home, and wanted to listen to the music, I pulled out the CD and put it in my Dad's CD player and got great audio. That was my choice at the time. Others may have real issues in the downgrade in audio I endured with the cassette reproductions from the CD. But, it worked FOR ME.

What works for you?

 

[Blue Dude]

I have all of my CDs ripped to FLAC because being lossless there will never be a concern over loss of quality and I like all the metadata options for library purposes. Delivering the content is easy and getting more painless all the time due to all the streaming apps available, but just copying files to my phone is already easy. For more critical listening I play the files with Kodi through my receiver.

And I remember dubbing vinyl albums to cassette back in the day so I could make them portable, and that was just fine. What a great time for music lovers. We've never had it so good in all of human history.