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Thread: AAC and WAV - Taste the difference.

  1. #21
    Join Date: Jun 2009

    Location: Yonkers, NY USA

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    If you decide to expand a FLAC file that was derived from a WAV or AIFF file, you will get back the exact same data as the original file, hence the definition lossless. That is a fact. Same with Apple Lossles, ALAC.

    The assumption, however, is that because the FLAC/ALAC file contains the same exact data, therefore it *must* sound the same.

    Does it? And if it doesn't, where is the flaw?

    Sad to say but bit perfect is only the beginning of truly good audio............

    CD
    David

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  2. #22
    Join Date: Jun 2010

    Location: Finland

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    Sorry Stan, but I don't believe that anyone can hear the difference between a WAV file (for example ripped from a cd) and a FLAC file converted from the WAV file (original). FLAC indeed is a lossless format as well as WAV; the recording devices (mics) however are not lossless.

    To prove that one can hear the difference between a FLAC and a WAV file, one should compare the original WAV file to a WAV file that has been converted from a FLAC file converted from the original WAV file. I believe that those too files are identical as digital music is only bits as mentioned before, and if the bits are exactly the same in both files, there is no difference between the files' audio content.

    Could you do this kind of ABX test to prove that there is a real difference that hasn't been caused by the extra processing power needed, software or by a placebo effect? I have never heard of anyone who could hear the difference.
    Janne

  3. #23
    Join Date: Jul 2010

    Location: North Cambs UK, Earth, Sol, Orion - Cygnus arm of galaxy

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    I'm MadeOfDeadGiantStarsThatExplodedEonsAgo.

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    I think what Stan is trying to say is that digital audio isn't lossless fullstop

    By definition an analogue signal has an infinite number of positions on a waveform whether that be in time or voltage. As soon as it has been converted to digital there are no longer an infinite number of voltages positions available, but a finite number & so many samples per second rather than an infinite number

    Therefore digital audio is lossy, some forms to a lesser degree but in the end it all is
    Bests, Mark



    "We must believe in free will. We have no choice" Isaac Bashevis Singer

  4. #24
    Join Date: Oct 2008

    Location: Southampton, UK and Nicosia, Cyprus

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    Quote Originally Posted by Reid Malenfant View Post
    Therefore digital audio is lossy, some forms to a lesser degree but in the end it all is
    Of course, and an analogue transmission system will degrade the music too, but in different ways. It therefore follows that an analogue transmission system is also "lossy".

    But we are stretching the meaning of "lossy" and "lossless" which is quite clear and given by context.

    That the digital compression and decompression employed in FLAC, ALAC and other "lossless" formats does indeed return the data to its original state ought to be a matter that can be objectively measured. I thought it had been proven that a FLAC file "unzipped" is identical to the file that was stuffed into the "zip". The process is not in principle different to zipping and unzipping other computer files.

    I cannot make a judgement about what Stan is hearing.
    Last edited by Labarum; 09-08-2010 at 16:03.
    Brian

    In Southampton: Raspberry Pi 4 running PiCorePlayer, Beresford Caiman SEG, Quad 77 Int Amp and CD Player, AVI Neutron 4, Sennheiser HD25 headphones.
    In Nicosia: Small Format HTPC, Beresford 7520 ,Quad 405-2, Quart 980s German Tower Loudspeakers.

  5. #25
    Join Date: Aug 2010

    Location: Montseny National Park, Catalonia

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    I'm John.

    Default Good Grief!

    Reid Malenfant.
    There is no “I think what Stan is trying to say” about it; hence my direct quote.

    I don’t think anyone would argue that sampling a wave form incurs a loss of data. The salient point is whether or not that lost data makes an audible difference.
    Dealing with the sampling of a waveform first;
    http://en.wikipedia.org/wiki/Nyquist...mpling_theorem

    The term “lossless” could perhaps be misleading for the exceptionally pedantic. However, most seem to realize that this term refers to what can be heard (quality), not to a data quantity. Of course there will always be those who say any sampling process, no matter what frequency the sample will give inferior sound quality, I recall being in that particular camp myself many years ago.
    What often seems to be forgotten is that the majority of Dacs also either over sample or up sample and that we have been waxing lyrical about expensive CD player’s sonic qualities for many years now happily listening to 16/44.1.

    That’s great StanleyB. They do say faith is half the battle
    However, no matter how hard I try to convince myself I can fly when I stand on that cliff edge with all the conviction I can muster science always rears it’s ugly head and says to me “John, don’t do it. What will happen is no matter how hard you flap your arms you will accelerate downwards at 9.81 metres per second per second until you reach your terminal velocity and spend the rest of the journey screaming your way to a messy and ugly death.”
    “After you StanleyB my good man” is all I can say to your doubts concerning scientific theory and a considerable amount of opinion gained through knowledge of the outcome of people threw themselves of cliffs.

    (Given the tone of Stanley’s post I hope I will be forgiven for a somewhat blunt response)

    With regard to your arrogant and blatant product advertising and your suggestion that I should buy one of your Dacs to test your assertion for myself I have to wonder why on earth I would want to replace the very competent Dac I have now with an inferior product?
    Single spur balanced Mains. Self built music server with 3 seperate linear PSU, Intel i5, 16 GB RAM no hard drive (various Linux OS). Benchmark Dac2 HGC, single ended XLR interconnects/Belkin cable. Exposure 21RC Pre, Super 18 Power (recap & modified). Modded World Audio HD83 HP amp. Hand built Monitors with external crossovers , Volt 250 bass & ABR, Scanspeak 13M8621 Mid & Scanspeak D2905/9300 Hi. HD595 & Beyer 880 (600 ohm) cans.

    The whole problem with the world is that fools and fanatics are always so certain of themselves, and wiser people so full of doubts.
    -Bertrand Russel

    John.

  6. #26
    Join Date: Feb 2008

    Location: http://www.homehifi.co.uk

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    Quote Originally Posted by Welder View Post
    Reid Malenfant.
    There is no “I think what Stan is trying to say” about it; hence my direct quote.

    I don’t think anyone would argue that sampling a wave form incurs a loss of data. The salient point is whether or not that lost data makes an audible difference.
    Why the sudden back tracking after banging your head against the wall ? I also noticed that you are now questioning if that loss of data makes an audible difference. Anyone else in stitches over their keyboard ?

  7. #27
    Join Date: Oct 2008

    Location: Southampton, UK and Nicosia, Cyprus

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    I think it's time to stop, Stan.
    Brian

    In Southampton: Raspberry Pi 4 running PiCorePlayer, Beresford Caiman SEG, Quad 77 Int Amp and CD Player, AVI Neutron 4, Sennheiser HD25 headphones.
    In Nicosia: Small Format HTPC, Beresford 7520 ,Quad 405-2, Quart 980s German Tower Loudspeakers.

  8. #28
    Join Date: Feb 2008

    Location: http://www.homehifi.co.uk

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    To stop what exactly Brian? Questioning the integrity of someone who implies I am a liar? The cock has not crowed three times yet...

  9. #29
    Join Date: Aug 2010

    Location: Montseny National Park, Catalonia

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    I'm John.

    Default The original point of the debate

    StanlyB.
    To be frank, I don’t think you have managed to follow the qualification and reasoning in my posts. There has been no “backtracking”. If you find my style of writing and reasoning hard to follow then I can only apologies and attempt to keep it simpler.
    The point in question is whether or not you or anyone else can tell the difference between a Flac file and a Wav file when played through the same medium.
    You say you can.
    Not only do you say you can but you say you can primarily because of the qualities of the Dac you make.
    However, it seems the vast majority of audiophiles can’t.
    In fact, you are the only person I have yet to come across that states they can hear such a difference.

    Attempting to gather forum support through being an established member having made a bit of a fool of oneself is a very old and generally despised tactic.

    If you are unable to stand up for yourself StanlyB using polite/humorous/intelligent/informed points of discussion then I suggest you pass this debate by.
    Single spur balanced Mains. Self built music server with 3 seperate linear PSU, Intel i5, 16 GB RAM no hard drive (various Linux OS). Benchmark Dac2 HGC, single ended XLR interconnects/Belkin cable. Exposure 21RC Pre, Super 18 Power (recap & modified). Modded World Audio HD83 HP amp. Hand built Monitors with external crossovers , Volt 250 bass & ABR, Scanspeak 13M8621 Mid & Scanspeak D2905/9300 Hi. HD595 & Beyer 880 (600 ohm) cans.

    The whole problem with the world is that fools and fanatics are always so certain of themselves, and wiser people so full of doubts.
    -Bertrand Russel

    John.

  10. #30
    Join Date: Apr 2010

    Location: Nergenshuizen, NL

    Posts: 197
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    I’m afraid I duplicate a bit what others has said before.

    Take a WAV file
    Convert it to FLAC
    Convert the FLAC back to WAV
    Load both in an audio editor, time align and subtract the 2 tracks.
    You end up with a track with zero’s only.
    Yes indeed, FLAC is lossless, just like WinZip or any other well implemented compression software.

    A slightly different test:
    - Record the SPDIF out in both cases
    - Load both in an audio editor, time align and subtract the 2 tracks.
    Again, you end up with a track with zero’s only.
    B.t.w. this has been done recently to test differences between iTunes and Amarra (playing of course both recordings at the same and native sample rate).
    This is probably the most convincing test as the SPDIF out is what goes into the DAC.
    All these test yield the same result, both files are bit identical.

    If 2 files are bit identical, can they sound different?
    PC based audio is PCM audio, you have the samples and the sample rate.
    Correct reproduction requires to things:
    - the right sample, any alternation to the bits will alter the sound
    - the right time, any unaltered sample arriving to early or to late will alter the sound

    Yep, the second one is known as jitter.
    A logical conclusion, as PCM audio consist of 2 components, both must be right.
    Any comparison of possible differences e.g. between audio formats should take both aspects into account.

    What happens if you play WAV?
    A codec will interpreted the content, translating it to raw PCM.
    As the content is almost identical to raw PCM, this codec has almost nothing to do so has very little impact on CPU use.
    However, as the file is uncompressed, you do have a lot of I/O

    What happens if you play FLAC?
    Exactly the same, a codec will interpreted the content, translating it to raw PCM.
    However, as the audio is compressed, it must be unpacked. This requires more CPU.
    The file is about half size compared with WAV, so I/O is substantially reduced.

    There are claims that FLAC sounds different and this difference is often qualified as worse than WAV.
    Do we have to conclude that CPU activity is more damaging than I/O?
    There are a lot of claims that I/O is detrimental to, one should use a memory player, loading the entire song in memory before playback starts.

    What would happen if we translate the WAV and the FLAC to PCM first, load the result in memory and start playback. Will they sound different?

    My personal opinion:
    DA conversion is a delicate process.
    Claims about the audible threshold of jitter varies but we probably talk nano or even pico seconds. http://thewelltemperedcomputer.com/K...fectJitter.htm
    I think it is perfectly possible that electrical activity going on inside a PC disturbs the clock of the sound card so maps itself into sample rate jitter.
    Increased system activity will decrease the sound quality.
    This is pretty much like having a video card and the more system activity, the more your screen starts to blur.
    One might argue that if sound quality fluctuates with system load, this indicates a design flaw.
    As a consequence, on a well designed system you won’t hear any difference and on the ones with a crappy sound card, you do.

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