# FLAC vs WAV which is better?



## mlundy57 (May 21, 2013)

I'm not exactly sure this is the right place for this question but here goes. For high definition music, 44.1Kz/24bit - 192Kz/24bit, which format is better, FLAC or WAV and why?

Thanks,

MIke


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## Kal Rubinson (Aug 3, 2006)

mlundy57 said:


> I'm not exactly sure this is the right place for this question but here goes. For high definition music, 44.1Kz/24bit - 192Kz/24bit, which format is better, FLAC or WAV and why?


No difference except if your particular system has its own issues with real-time decompression of the FLAC files.


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## mlundy57 (May 21, 2013)

Is anything lost when converting between files? I have been downloading HD audio files in FLAC. Three of my units (the ones I listen to most) can read the FLAC files. However, two others cannot. For these I use J River to convert the FLAC files to WAV files. Does this cause any degradation?

Thanks,

Mike


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## jimbodude (Jul 26, 2011)

If you use the FLAC lossless compression, then there is no possibility of losing any resolution. It is a mathematical property of the encoder and decoder. Be careful - FLAC does have a lossy option - sometimes called "near lossless" - which will cause some loss compared to the original. Whether the loss is audible is a topic with much subjectivity and complexity.

You are completely safe using FLAC lossless to save some space, so long as your tech can play it back. Additionally, you are guaranteed that an output WAV created from a lossless FLAC is exactly the same as the original source, assuming the encoder and decoder are implemented properly - which is a safe assumption.


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## mlundy57 (May 21, 2013)

Thanks


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## AudiocRaver (Jun 6, 2012)

Interesting trend: The direction used to be toward flac (or - ugh! - MP3) to save storage space. Today HD and memory space are so cheap the trend appears to be going the other way, toward true hi-rez 24/96 and 24/192 remastered files. Why compress any more?


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## mlundy57 (May 21, 2013)

Now they are talking about DSD for ultra high resolution digital files. You need a DSD DAC to play these files as DSD However, J River Media Center 18, and probably other digital music players as well, can read DSD files and convert them to PCM for playback. Oppo has a firmware upgrade for the BDP 105 so it can read and play DSD files directly.


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## gdstupak (Jul 13, 2010)

A benefit of FLAC is that it's ID tagging is more widely accepted.

Also, FLAC now offers an *uncompressed* lossless format, so no chance of decompression artifacts. I recently converted all of my thousands of WAV files into FLAC uncompressed/lossless for use with JRiver MediaCenter.


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## AudiocRaver (Jun 6, 2012)

Interesting. The likelihood of decompression errors from hard disk storage seems like in the realm of processor error rates, extremely low. There are also indications of decompressing/streaming flacs sounding _different_ from streaming wavs, supposedly from the added complexity of the decompression process and - what? - more jitter? Not sure what the possible explanations might be.

HOWEVER...

Disk space is cheap. Simpler is better. Tags are handy. Uncompressed FLACs sound like the way to go.


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## RTS100x5 (Sep 12, 2009)

I use JRIVER output options to convert all output to 5.1 / 96k / 24bit and life IS wonderful (is there any other media player that will do that??) ... However I do use ADOBE AUDITION to batch convert most of my audio from MP3 (from Amazon) to 96k / 24bit....and it will also encode it to a 5.1 surround sound though Im not certain what format this "surround sound" is, other than your options for bitrate and file type - and Lossless WAV is 1 of those options so I use that....


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## AudiocRaver (Jun 6, 2012)

RTS100x5 said:


> I use JRIVER output options to convert all output to 5.1 / 96k / 24bit and life IS wonderful (is there any other media player that will do that??) ... However I do use ADOBE AUDITION to batch convert most of my audio from MP3 (from Amazon) to 96k / 24bit....and it will also encode it to a 5.1 surround sound though Im not certain what format this "surround sound" is, other than your options for bitrate and file type - and Lossless WAV is 1 of those options so I use that....


What is the advantage of converting MP3's to 24/96 WAV format?


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## RTS100x5 (Sep 12, 2009)

AudiocRaver said:


> What is the advantage of converting MP3's to 24/96 WAV format?


the overall sound quality is much better :yes: .but mostly I do conversions to share music and my car stereo wont play MP3...


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## Kal Rubinson (Aug 3, 2006)

RTS100x5 said:


> the overall sound quality is much better :yes:


Not sure how you intended the smiley but really?


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## RTS100x5 (Sep 12, 2009)

Really the sound quality is better


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## AudiocRaver (Jun 6, 2012)

RTS100x5 said:


> Really the sound quality is better


I am trying to be open-minded here. When a music file is converted to MP3, information is lost, no way to get it back. An MP3 converted to FLAC does not - can not - recover that information. Playing that FLAC presents exactly the same data series, thus the same audio stream, as playing the MP3 version would do. I do not see how there can be a difference in audio quality.

To what do you attribute the improvement in audio quality?

Edit: Also, you mention sharing files. Most - really ALL - people who look for shared FLAC or 24/96 files want them for their pristine quality, and would be disappointed to find they were converted from MP3. Not advocating sharing, of course, just stating what people are looking for.


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## RTS100x5 (Sep 12, 2009)

Rather than trying to explain it I suggest trying it yourself.... true information cannot mathematically be recovered but existing information can be enhanced with current technology....


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## morca (Aug 26, 2011)

I tested whit MP3 cd/WAV and hires files
MP3 has missing pieces ,what you do,you let JRIver upsample/fill in the pieces to hires.
SQ is still like the MP3 over here.

Also tested cd/WAV upsample using foobar/secret rabbit to 192-24,the SQ does not get better,beter sad worse.

I think you mean there is more BOOM,but the SQ/detail does not come back whit an upsample.


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## AudiocRaver (Jun 6, 2012)

The real question becomes: Does up-sampling make any file sound better? If so, how much, how often, and under what circumstances? And can it be verified with careful A-B testing? Even determining just how to go about that testing is not a trivial thing. Again, I am trying to be open-minded, although admittedly a bit skeptical.

A related question relevant to this thread: Is there an audible difference between a WAV and a properly encoded FLAC of the same song? Some claim that there is. If so, how much, how often, and under what circumstances? And can it be verified with careful A-B testing? This one I am even more skeptical of, but at least it should be pretty easy to set up A-B testing to verify.


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## erwinbel (Mar 23, 2010)

AudiocRaver said:


> The real question becomes: Does up-sampling make any file sound better? If so, how much, how often, and under what circumstances? And can it be verified with careful A-B testing? Even determining just how to go about that testing is not a trivial thing. Again, I am trying to be open-minded, although admittedly a bit skeptical.
> 
> A related question relevant to this thread: Is there an audible difference between a WAV and a properly encoded FLAC of the same song? Some claim that there is. If so, how much, how often, and under what circumstances? And can it be verified with careful A-B testing? This one I am even more skeptical of, but at least it should be pretty easy to set up A-B testing to verify.


I haven't heard of "un-properly encoded FLAC"...

Many claim to hear differences between FLAC and WAV and/or between AIFF and Apple Lossless. But I have yet to read about the first double blind listening test that proves that claim!

Almost all my music is Apple Lossless. I did test against AIFF and WAV but couldn't hear a bit of difference. The only exception I make are 24-bit / 192 KHz files, since some devices don't play Apple Lossless 24/192 (only up to 24/96). 

I am waaay to about album art and correct tagging to even consider WAV.

PS: it's obviously 100% wrong to upsample MP3 to Lossless since MP3 is Lossy and once something is lost, it's gone for ever.


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## AudiocRaver (Jun 6, 2012)

erwinbel said:


> I haven't heard of "un-properly encoded FLAC"...


I am not sure how it happens. The FLAC encoder has an option as it encodes to FLAC of having another parallel process running which decodes back to WAV and compares to the original file, so any encoding errors are caught immediately. But I do not know how or why en encoding error ever _would_ happen. It would be like having a calculator program tell you that 10 + 10 = 19. How often do you see that?

But apparently it happens somehow. I was recently converting a bunch of files from FLAC to WAV and there were a couple that would not decode. They somehow had been corrupted and the FLAC processor could not decode them. Somehow they had become "un-properly encoded FLACs." Seems like it would be as rare as a processor error or memory read error.



> Many claim to hear differences between FLAC and WAV and/or between AIFF and Apple Lossless. But I have yet to read about the first double blind listening test that proves that claim!
> 
> Almost all my music is Apple Lossless. I did test against AIFF and WAV but couldn't hear a bit of difference. The only exception I make are 24-bit / 192 KHz files, since some devices don't play Apple Lossless 24/192 (only up to 24/96).
> 
> ...


I have heard someone say in a post that they were aware of a double-blind study where the difference was audible, but I have not seen such a study myself or heard of it from an authoritative source. I believe it would be indicative of a device (cd player, media player, etc.) of very poor quality that the extra workload of decoding the FLAC would cause noise somehow.

No doubt there are plenty who claim to be able to hear the difference. I am doubtful, but then I have not tried to yet.


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## gdstupak (Jul 13, 2010)

AudiocRaver said:


> The real question becomes: Does up-sampling make any file sound better?


Upsampling can change the sound of audio by moving bad artifacts. It shifts preringing into postringing which is supposed to be less noticeable. I haven't tried, so I can't give any personal input here.

As far as adding resolution (info) with upsampling: If a video scaler can make video look better and fuller by adding info, why can't it be done with audio?

Here is an article from Sound & Vision that gives better detail about upsampling for DolbyTrueHD movies (Geoffrey Morrison says... "_*Despite warnings by Dolby that the effect was subtle, I found it readily apparent, as did my colleagues also present at the event. This is not the sort of change in sound like going from MP3 to CD, or even from CD to high-rez. The level of change is more like switching to a better DAC: subtle, but noticeable*_" :

http://www.soundandvision.com/content/new-format-news-dolby-truehd-advanced-96k-upsampling


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## lcaillo (May 2, 2006)

Let's clarify this notion about information being added or not with upsampling and what upsampling is. 

In a digital representation of some signal, each sample represents an average of some amount over some period of time. The original information from that period that is not equal to the value of the sample is lost in terms of representation by they digital signal. It will never be transmitted through the system. What can be done is estimating what might have been there and inserting that information into a higher sample rate representation. There are many ways to do the estimating and to produce an upsampled result. Whether the result represents an improvement or not is just like any other process, it depends on the how well the information is predicted and how well the resulting signal is assembled.

To say "upsampling" is good or bad is like saying apples are rotten or not. Unless you know more about the process, it is impossible to generalize. 

As for moving artifacts forward or backward in time, there should be none of that and it depends as much how the D/A conversion is done. Might there be such a shift? Certainly. In the digital transformation, however, there is no reason not to account for that, any more than phase shifts in digital filters in general. But again, the D/A has much to do with what artifacts will be produced and how.

There are so many variables in an upsampling discussion that it is one of the best examples of my mantra...it depends. I have seen some very clever predictive algorithms and uses of machine learning systems to recreate signals from lousy or noisy samples. I have no doubt that one might be able to produce a "better" version through upsampling in many cases. There is just so much that can't be captured in one word " upsampling" that to say it isbetter or worse is impossible as a generalization.

We have to talk very specifically about what is going on in the process and do some very careful testing, and the results are likely going to vary with material. Any upsampling process is a statistical guessing game. How well the guesses are informed may not be a constant across all kinds of signal conditions, and music and video represent very variable and complex conditions.


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## Greenster (Mar 2, 2013)

I have been reading about Dolby TrueHD lately. 

http://www.dolby.com/us/en/consumer/technology/home-theater/dolby-truehd.html

It is able to up sample at 96khz which is well above what a human ear can hear but the real question may not be what we can hear at those upper frequencies but what we can feel. Also the upper in audible frequencies are able to make music sound fuller. 

http://www.soundandvision.com/content/new-format-news-dolby-truehd-advanced-96k-upsampling

Very interesting read. I think a ton will have to do with the speakers as well as the room. 

That said, I can tell a difference between FLAC files vs. mp3's. But as far as FLAC vs Apple lossless or Wave files. Well they are just too close to each other by what is missing in the file to really tell any difference to me.


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## tonyvdb (Sep 5, 2007)

Greenster said:


> That said, I can tell a difference between FLAC files vs. mp3's. But as far as FLAC vs Apple lossless or Wave files. Well they are just too close to each other by what is missing in the file to really tell any difference to me.


I just recently re ripped all my 600 or so CDs and experimented with the different types of file formats. I used my Lee Ritenour 6string theory as my test CD because it's one of the best dynamic recordings I have heard to date.
Agreed that the difference between FLAC and apple lossless was impossible to tell where I was surprised was using a VBR setting at very high using 256kbs I was also unable to hear any change unless I put my AKG headphones on and even then it was so subtitle I'm not sure I was just thinking I did. The smaller file size was a big decision maker for me so I have used the VBR 256kbs setting for all.


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## AudiocRaver (Jun 6, 2012)

tonyvdb said:


> I just recently re ripped all my 600 or so CDs and experimented with the different types of file formats. I used my Lee Ritenour 6string theory as my test CD because it's one of the best dynamic recordings I have heard to date.
> Agreed that the difference between FLAC and apple lossless was impossible to tell where I was surprised was using a VBR setting at very high using 256kbs I was also unable to hear any change unless I put my AKG headphones on and even then it was so subtitle I'm not sure I was just thinking I did. The smaller file size was a big decision maker for me so I have used the VBR 256kbs setting for all.


In my experience, MP3 loss is more readily detectable with a recording that has cymbals, bells, triangle, sometimes a female vocal with clean, strong sibilants... sustained complex HF content. To me those sounds are _grainy_ and _messy_ with MP3s, worse at lower bit rates.


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## tonyvdb (Sep 5, 2007)

AudiocRaver said:


> In my experience, MP3 loss is more readily detectable with a recording that has cymbals, bells, triangle... sustained complex HF content.


Agree, my old files were mostly ripped at 160kbs fixed bitrate and I could hear a big difference in the highs when I went up to the new settings. My EVs are not a forgiving speaker and flaws in the recordings show up very quickly. The imaging also opened up with the higher bitrate.


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## wd3 (Feb 23, 2014)

I use FLAC files because they are lossless, still are compressed to a degree, easily decompressed in hardware, and standardized without a specific affiliation to a specific OS company (i.e. Microsoft or Apple). The last point is important for me due to portability. Most devices (i.e. Bluray players, Recievers, and etc.) and various applications now support FLAC. I'm also a big user of Linux at home which supports FLAC very well, and there are a variety of applications for Mac or Windows users too. As far as sample rates are concerned, I think you would be hard pressed to actually hear any real differences in a scientific double blind study between 44.1khz/48khz and higher sample rates (88.1, 96, and 192). Keep in mind that the Nyquist theorem states to accurately sample any signal it only needs to be sampled at 2 times the signal frequency. This is the basis for all digital music. Also keep in mind that as we age, our hearing will dimension over time. In fact, at present I'm in my mid forties, and I don't hear anything over 13.5 to 14khz which is nothing unusual for a person my age. Although, that means that 44.1khz/48khz is already providing 3-4 times or more the sampling rate needed to accurately represent the signal. So, by increasing the sampling rate it will probably only add disc space requirements and potential artifacts if the algorithm isn't done correctly. In fact, by upsampling you are basically replicating a typical component already used in higher end DACs. For instance a DAC that has an Asynchronous Sample Rate Converter (ASRC), DAC, and opamp chain will resample any incoming signal to an optimal sample rate for the DAC chip. This is typically around 100 khz or so. Some use a higher rate and some use a lower rate. These are done in hardware and with a very complex mathematical formula. Can the computer do that? Theoretically, that should be no problem, but in practice most programmers haven't replicated the complex math completely. In addition, it requires a lot more CPU power than you would like especially if doing it on the fly for playback now. So, they make compromises to allow their software to run on most platforms (read: designed to the lowest common denominator because you don't want high cpu usage and a long time for conversion especially on slower computers). Also, that means that if you already upsampled the data, that it will be resampled to another probably lower rate. For instance, if you upsample to 192khz, your ASRC might just resample it down to 110khz or whatever is optimal for the DAC chip in use. In short, I don't believe upsampling will always give you a good result. For higher bit depths, I think an argument can be made for its benefits. I don't think over 24bits would be useful, but even 16bits has such a low noise floor that I think again it would be difficult if not impossible to discern any difference in well recorded recordings using a double blind scientific study. So, if you want the best quality audio files using today's technologies, my recommendation would be to obtain CDs and use a software package to convert them to 44.1khz/16bits using a lossless format such as FLAC. That software should also read the CD multiple times to handle error correction because CDs do not have redundant data like most data discs. It can also be advantageous if that software double checks its database for multiple pressings to confirm CD accuracy. Or you should get audio files from reliable sources (that get them from the studios) in a format like FLAC in 44.1khz/48khz with a 16 bit or 24 bit depth. There have been some scientific studies to show that frequencies such as 96khz and 192khz can actually harm audio quality depending on the playback equipment used. Also, keep in mind that once the audio stream has been run through a lossy algorithm such as MP3, you can never get the data back. MP3 was designed to allow for small storage, and it uses psychoacoustic algorithms to convert an audio stream to something that approximates the original. So, in most cases you can't convert that back into a lossless format like FLAC and get the missing components. Just like you can't convert a 16bit file to a 24bit file and gain anything from that.


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## talmadge (May 4, 2010)

My vote: FLAC


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## Savjac (Apr 17, 2008)

Having played with most of the different ways to save music, I can say, in my experience, FLAC seems to be the way to go, nice size, no sonic losses and you get to keep the metadata information. No downside here.


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## talmadge (May 4, 2010)

Agree


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## NBPk402 (Feb 21, 2012)

Greenster said:


> I have been reading about Dolby TrueHD lately.
> 
> http://www.dolby.com/us/en/consumer/technology/home-theater/dolby-truehd.html
> 
> ...


Wouldn't 96khz be an overtone... If so I would think that it might effect the sound of the original frequency if it was cur or altered.


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## AudiocRaver (Jun 6, 2012)

Another vote for FLACs. I usually use wave files for my evaluation tracks just to eliminate questions. But FLAC is definitely the way to go for general use.


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## AudiocRaver (Jun 6, 2012)

ellisr63 said:


> Wouldn't 96khz be an overtone... If so I would think that it might effect the sound of the original frequency if it was cur or altered.


Good question. Changing the amplitude of an overtone of a piano note, for instance, will only change the sound of that piano note if the overtone is audible to the listener. If the listener cannot hear that overtone, a change to its amplitude will not affect that sound in any way.


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