quality flac files
Jan 27, 2013 at 9:53 PM Post #17 of 38
Well a 16-bit 44.1KHz FLAC file is nearly perfect when paired with a high-quality DAC by human hearing standards. It's lossless so compression uses algorithms to compress, and has no effect on sound quality (the option exists so slower processors can encode and decode easier/faster). The one thing that you should watch out for is remastering quality, this has a major effect on sound and is the reason so many people are switching back to vinyl (vinyl costs more to press than a CD so usually higher mastering standards are put in place).
 
Jan 29, 2013 at 2:31 PM Post #19 of 38
Quote:
 
Spectral analysis is an excellent way to check to see if the copy you have isn't really an .mp3 or some other nonsense under a different name.
 
http://www.whatinterviewprep.com/prepare-for-the-interview/spectral-analysis/

Do you know software that automatically does such an analysis and that can tell with high probability if the file is a genuine FLAC or not?
 
Jan 29, 2013 at 4:26 PM Post #22 of 38
Quote:
 
No. But it's a really easy skill to learn, if you really care to do it.

I know, but if there are more files to check such an application would prove useful. 
Actually like two years ago I found a little application (forgot the name) that was supposed to do it (saying the probability in percentages), but it turned out to be quite unreliable. Maybe during last two years someone made better software for this purpose. I might look in my free time.
 
Jan 29, 2013 at 5:19 PM Post #24 of 38
Quote:
Great recording
Great re-masters
There is a difference and  you have to pay for it.
https://www.hdtracks.com

 
Paying a premium for audio tracks guarantees nothing except increased expense.
 
HDTracks isn't a recording or mastering studio. They simply sell what they're provided. Analysis of some of their tracks (on another web site) shows that their audio files are no better and no worse than any other random source. It's not a requirement to pay extra for high-quality audio but people are free to spend their money as they please.
 
Jan 29, 2013 at 5:26 PM Post #25 of 38
Note:   I got lazy and I didn't want to proof read this.......  Hopefully you will get the gist of My Opinion.
 
I don't know why I am posting on this topic because it usually gets very heated when discussing music file processes, formats and "quality".  I know I'm going to get my chops busted, so here it is, my one and only post.   But I am all for being corrected when I muck up the facts....

FLAC, ALAC, AIFF, MP3 are all music / media file types.  How the bits are assembled into a file.  Just as .doc is a Word file.xls is an Excel file, .jpg is and image file and .raw / .nef is a raw image file
 
Converting from one format to another will not change the data.  In fact, it might limit it.  Save an Excel file as a .pdf and you can't edit the .pdf, however a browser can now see the file as an image.   iTunes can't read FLAC so it needs to be converted to a format it can read, AIFF or ALAC.
 
IMO, the quality, or sonic complexity of a recording is based on a couple things.  First the analog stuff: The studio, as opposed to say live, the recording equipment mice, tape machines, how the tracks are mixed or mastered.  Then the digital stuff.
 
At this point the analog recording gets picked up by the ADC  (Analog to Digital Converter).  The ADC "samples" the analog files it picks up.  The microphonic vibrations are sampled at "X times a Second"  This is "Hz" or a "cycle" or a "Clock rate".   The standard rates are 
44,100  88,200  96,000  176,000  and 192,000   this is how many time each second the ADC "samples" the analog source.  The samples are assembled into either 16 bit or 24 bit "bundles"
 
The recordings are would then be differentiated by a combination of these factors.   Remember Hz hertz is times per second and kHz is 1000 times a second.   So a recording could be 44.1kHz/16   or samples 44,100 times each second and assembled into 16 bit bundles.
Another might be 192 kHz/24    192,000 x per sec "sample rate" into 24 bit "bundles"
 
Then the files are encoded into ….. Mp3, 320 Mp3, FLAC, ALAC, etc….   The key consonant is "L"  That means lossless.   (Let the bashing begin…)  You get all the bits that the ADC picked up from the analog source.   If the sample rate was 88.2kHz twice as many samples were taken than a 44.1kHz rate.   The file is bigger.  Most of the tracks I purchase from HD Tracks are over 90MB per track.  An iTunes Mp3 is about 6MB.   Same song.  Media files are not like a .zip file where you decompress the file after you load it.  You DAC does not decompress files.   You are listening to what the encoding algorithm re-represented the sample bits to be.  The "low hanging fruit" for a compression algorithm is repetitive data like silence.   It picks up a bunch of sequential silence and represents it as a couple of bits.
 
 I am a photographer, I like to use the analogy of .jpg versus .raw files.  It's the same thing.  The camera sensor, when set to .jpg, re-represents what it picks up and writes it to a reduced file.  Big boy cameras record every bit the sensor picks up and writes it to a "raw" file.   Compression is compression and the camera makes a great analogy to music.  IMO  IMO  IMO
 
When you zoom 400x or 800x into a RAW image you can see a finer matrix of pixels.  More important you can see the "Transitions" of color between between the matrices.  A jpg image has far fewer pixel components, there are fewer transitions to work on. Hence you  touch up the image with anywhere near the accuracy of a RAW file.  Prints from RAW have much more depth.  If you google an image there will no double be duplicates of the same image in multiple resolutions.  If you enlarge them the resolution of the smaller image will break up as it is enlarged.
 
IMO, where music compression  also breaks down is at the edges or the transitions.  This is visible in an image in photoshop. 

 It's harder to differentiate in music.  But you may detect "something" is different.  Yes.. fuller…. more complex perhaps.  I feel like I am in the recording studio standing in the middle of the instruments.  Drum back left, guitar front right, vocals middle.   I can detect the "transitions", the falloff of note to silence in the HD recordings.
 
The recording and mastering is without a doubt the most important factor.  After that. I'd have to give it to sample rate, then encoding.
 

People will say the human ear can't tell the difference.  I guess I have great hearing because I definitely detect "A" difference.  In short.  I want all the music that was picked up in the studio so I can listen to the parts I am able to hear.  Maybe I can't hear all of it, but perhaps more than what the compressed version lets me hear.  I want to make that decision.

 
Jan 29, 2013 at 5:56 PM Post #26 of 38
Quote:
 
Paying a premium for audio tracks guarantees nothing except increased expense.
 
HDTracks isn't a recording or mastering studio. They simply sell what they're provided. Analysis of some of their tracks (on another web site) shows that their audio files are no better and no worse than any other random source. It's not a requirement to pay extra for high-quality audio but people are free to spend their money as they please.

Yep,
I hear ya.
HD Tracks only sells albums, they don't record them.
 
That said, they get their content Directly from over 100 Recording Labels rather than "random sources"
https://www.hdtracks.com/index.php?file=labellist
 
The only source I want my music from is the record label and an approved distributor.  (iTunes, Google Play, Amazon, HD Tracks, Linn, etc.)
It's the record label's reputation that HD Tracks is delivering the sources that they are representing to the buyer.  It would be fraud to sell a track with 80MB of filler "0"s
 
Jan 29, 2013 at 6:01 PM Post #27 of 38
Quote:
Do you know software that automatically does such an analysis and that can tell with high probability if the file is a genuine FLAC or not?

 
Use Audacity and load up the FLAC and then do Analyze > Plot Spectrum. If the audio is cut off at 20kHz then its likely a FLAC created from an mp3. If there is data above 20hz then its probably a legitimate lossless file.
 
EDIT: This method doesn't always work. I use Easy CD-DA Extractor to convert FLAC files to mp3s for portable use and in Easy CD-DA Extractor I have the high and low pass disabled which results in mp3 files that do in fact have data over 20kHz.
 
Jan 29, 2013 at 7:41 PM Post #28 of 38
Quote:
Quote:
Great recording
Great re-masters
There is a difference and  you have to pay for it.
https://www.hdtracks.com

 
Paying a premium for audio tracks guarantees nothing except increased expense.
 
HDTracks isn't a recording or mastering studio. They simply sell what they're provided. Analysis of some of their tracks (on another web site) shows that their audio files are no better and no worse than any other random source. It's not a requirement to pay extra for high-quality audio but people are free to spend their money as they please.

 
The issue where some of their high-res albums turned out to be up-sampled CD quality has been resolved. They now immediately deal with any albums that turn out to have issues with their encoding.
 
Replies inline for the next one. There is some incorrect information in it.
 
Quote:
Note:   I got lazy and I didn't want to proof read this.......  Hopefully you will get the gist of My Opinion.
 
I don't know why I am posting on this topic because it usually gets very heated when discussing music file processes, formats and "quality".  I know I'm going to get my chops busted, so here it is, my one and only post.   But I am all for being corrected when I muck up the facts....

FLAC, ALAC, AIFF, MP3 are all music / media file types.  How the bits are assembled into a file.  Just as .doc is a Word file.xls is an Excel file, .jpg is and image file and .raw / .nef is a raw image file
 
Converting from one format to another will not change the data.  In fact, it might limit it.  Save an Excel file as a .pdf and you can't edit the .pdf, however a browser can now see the file as an image.   iTunes can't read FLAC so it needs to be converted to a format it can read, AIFF or ALAC.
 
An MP3 removes data, so should be removed from this list. AIFF and WAV are raw data file types. ALAC and FLAC are compressed (e.g.: rather like zipping a regular file).
 
IMO, the quality, or sonic complexity of a recording is based on a couple things.  First the analog stuff: The studio, as opposed to say live, the recording equipment mice, tape machines, how the tracks are mixed or mastered.  Then the digital stuff.
 
At this point the analog recording gets picked up by the ADC  (Analog to Digital Converter).  The ADC "samples" the analog files it picks up.  The microphonic vibrations analogue waveform is are sampled at "X times a Second"  This is "Hz" or a "cycle" or a "Clock rate".   The standard rates are 
44,100  88,200  96,000  176,000  and 192,000   this is how many time each second the ADC "samples" the analog source.  The samples are assembled into either 16 bit or 24 bit "bundles"
 
16-bit and 24-bit refer to the number of possible values available for each sample.
 
The recordings are would then be differentiated by a combination of these factors.   Remember Hz hertz is times per second and kHz is 1000 times a second.   So a recording could be 44.1kHz/16   or samples 44,100 times each second and assembled into 16 bit bundles.
Another might be 192 kHz/24    192,000 x per sec "sample rate" into 24 bit "bundles"
 
Then the files are encoded into ….. Mp3, 320 Mp3, FLAC, ALAC, etc….   The key consonant is "L"  That means lossless.   (Let the bashing begin…)  You get all the bits that the ADC picked up from the analog source.   If the sample rate was 88.2kHz twice as many samples were taken than a 44.1kHz rate.   The file is bigger.  Most of the tracks I purchase from HD Tracks are over 90MB per track.  An iTunes Mp3 is about 6MB.   Same song.  Media files are not like a .zip file where you decompress the file after you load it.  You DAC does not decompress files.   You are listening to what the encoding algorithm re-represented the sample bits to be.  The "low hanging fruit" for a compression algorithm is repetitive data like silence.   It picks up a bunch of sequential silence and represents it as a couple of bits.
 
See above about the difference between MP3, which removes data, and FLAC and ALAC where the data is simply compressed. The computer de-compresses it and sends it to the DAC as PCM data.
 
 I am a photographer, I like to use the analogy of .jpg versus .raw files.  It's the same thing.  The camera sensor, when set to .jpg, re-represents what it picks up and writes it to a reduced file.  Big boy cameras record every bit the sensor picks up and writes it to a "raw" file.   Compression is compression and the camera makes a great analogy to music.  IMO  IMO  IMO
 
I don't have the link handy but the way cameras save JPG data is radically different to the way the RAW data is handled.  The only similarity to audio is the amount of data being removed is considerable, but the compression itself is completely different.
 
When you zoom 400x or 800x into a RAW image you can see a finer matrix of pixels.  More important you can see the "Transitions" of color between between the matrices.  A jpg image has far fewer pixel components, there are fewer transitions to work on. Hence you  touch up the image with anywhere near the accuracy of a RAW file.  Prints from RAW have much more depth.  If you google an image there will no double be duplicates of the same image in multiple resolutions.  If you enlarge them the resolution of the smaller image will break up as it is enlarged.
 
You can't zoom 400x or 800x into a RAW image. We don't live in the future like you see in the movies where you can zoom endlessly into pictures. I think you mean 400% or 800% on your screen, which is far lower resolution than the image from a good, modern camera.  A jpeg file has the same number of pixels as a RAW image but the data is compressed to a degree that the finer detail of the image is often lost, as well as some colour information that makes fine tuning of the file's colours impossible.
 
[snipped]

 
Jan 29, 2013 at 9:40 PM Post #30 of 38
Quote:
What do you all think of dithering? It is a digital/analog filter which smooths the sine wave so that it is less blockey and has less harmonic distortion (at the cost of a higher noise floor).

 
I read something somewhere that convinced me that it was unnecessary, but I forget where. Source amnesia is at play here.
 

Users who are viewing this thread

Back
Top