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14K views 38 replies 7 participants last post by  soundandfury 
G
#1 ·
Re the RCO free down loads that we all must have tried by now. These are the first d/l that sound ok when put on CD and played on my HiFi system, I had given up with getting decent results from the net. I would guess the d/l rate of 320 kbps is the reason
This now poses a couple of questions for me:

1. The splitter that I am using "Wave Pad Sound Editor" can convert to a heck of a lot of formats I have chosen WAV under the impression that this is the best to use,
using Windows Media Player I am told the RAW format will be the default format,
So is WAV nearer to the original than RAW ??

2. What format do the pros use on the commercial CDs ??

3. Can you decompress the d/l [mp3] to get the original "Pro CD" format, I realise that each transfer looses a bit of the quality
 
#2 ·
Hi Andante

Let's go through your points one at a time.

1. WAV is short for waveform, and CDs use what's called PCM (Pulse-Code Modulation) WAV. But you don't need to worry about such technical stuff. All you need to know is that WAV is essentially the sound, as it is (i.e. uncompressed) on a CD.

Now, I've never come across RAW, so I can't comment on this, but what I will say is that converting lossy formats like MP3 (it's called 'transcoding', BTW) which have had digital information stripped out in the encoding process is not a great idea. Far better to just let your chosen program make an audio (as opposed to a data) CD, and accept that while the end result won't be up to the quality of a shop-bought disc, it'll nonetheless be as close to this as you can get given that you're making the CD from 320kbps MP3 files.

2. I've answered this in the above point: PCM WAV.

3. In short, no. You can NEVER recover lost information from a lossy format like MP3 or WMA. All you can do is burn these files as a standard audio CD and enjoy an approximate replication of what the original CD sounded like.

Hope that helps a little. :)

FK
 
G
#3 ·
Thanks Kuhlau I should have answered earlier but I am still trying to work it all out, with the better results obtained with the 320kbps I wonder why it is not used more for classical d/l, even allowing for the longer time involved it would serve the receiver better but may be a pain to the sender?
 
#4 ·
Many download sites are now switching to 320kbps MP3 files. It gives them the credibility to claim 'highest quality' (for the format, anyway), and is a convenient compromise between storage space requirements on their servers and relatively quick download times for the majority of customers.

Personally, however, I prefer FLAC these days. Not only does this technology allow EXACT replication of the digital information on a CD by using an intelligent compression method that reduces filesizes to about 60%, but it also eliminates the gapless playback issues that have long plagued the MP3 format, and which seriously impact on the enjoyment of classical music in particular.

FK
 
G
#5 ·
Please enlighten on FLAC sounds too good to be true what is the catch??
regarding the gap less play back I have the Wave Pad Sound Editor which allows you to make the division between movements but I am still learning how to refine it lol.
 
#6 ·
FLAC stands for Free Lossless Audio Codec ('codec', incidentally, is techie shorthand for 'encoding', or making the compressed format, and 'decoding', or 'unpacking' it, so to speak, ready for playback).

Best advice would be to Google the following terms, read carefully, experiment and come back here with any queries:

FLAC

EAC (Exact Audio Copy)

foobar2000

MediaMonkey

Winamp


The last three terms are all names of media players which can handle - and create - FLAC files. EAC is the best and most accurate freeware music ripper in the business. And FLAC ... well, you'll see. ;)

Good luck!

FK
 
G
#7 ·
Just to get one thing clear, if I run one of the RCO 320kbps d/l through say MediaMonkey
(which I have done as FLAC but not yet burned to CD) will it convert the mp3 to FLAC? I have noticed that the file size has doubled, I ask this because you said ""In short, no. You can NEVER recover lost information from a lossy format like MP3 or WMA."" :confused: :confused:
 
#8 ·
The important thing to note is this: an MP3 loses information during the encoding process which can never be recovered. If you want true FLAC files, you need to create these by ripping and encoding from the source CDs.

What typically causes confusion for those new to all these technologies is, as you've discovered, that filesizes inevitably increase during any transcoding exercise. Although sizes do get bigger, this doesn't, however, indicate that information originally lost has been recovered (sorry). :(

FK
 
G
#9 ·
So we can never improve the audio quality of the mp3 down load ??
A friend has requested that I ask you this question : Where does the extra information on the CD come from (larger file size) when you decompress the compressed file format and why does it sound much better when you do so?  :)
 
#10 ·
The 'extra information' is simply a result of burning an MP3 file to CD. I don't know the technical ins and outs, but I do know that despite whatever your ears may tell you, an MP3 file burned to a disc is IDENTICAL in terms of its sound quality.

To prove this point for yourself, make an MP3 file of a piece of music using whatever program you like, and make sure the encoding bitrate is 96kbps (which is very low quality). Next, burn that file to a CD, then listen to both the MP3 file you created and the CD burned from it.

Notice anything? You should be able to hear clearly that both the MP3 file and the CD sound equally poor. But - and this is the curious bit - look at how big the CD filesize is after this low-quality MP3 file has been transferred to disc. Still large, isn't it? Yet the quality is, as I say, identical to that of the MP3 file from which it was burned.

Don't trouble yourself too much about this. There are those who can't tell the difference between a 128kbps MP3 file and the source CD. There are also those who can clearly hear compression artifacts in a 320kbps MP3 file. We're all different in terms of how we perceive sounds, it seems.

FK
 
G
#11 ·
This is his reply, and I will make it the last as it is getting beyond my understanding:)

Don’t want to get too technical either – but basically when audio is stored in digital form the signal is broken down or converted into bits of information – these bits can be reconverted into the analogue signal. The formats that result from converting the analogue signal into digital uses various algorithms – these are often “proprietary brands of formats”, each one using different algorithms (mathematical sets of rules) – there are also different ways and algorithms for the processing of the digital information in the CD player – you may remember the debate about single bit vs. multibit processing – but what we are talking about here is the digital storage format of the music.
Now the more attributes of the music you store of the original music, the bigger the storage file.
A .wav file stores a large number of attributes and therefore is high quality. To save on space mainly but also I guess to promote your unique brand of software various formats with different processing of the original .wav high quality digital information have evolved. Generally they compress the information or leave out some attributes of the original digital file and make it smaller.
If you have a high bitrate it generally means a lot of attributes of the music is stored in the digital file
Lower bitrate is lower quality signal.

Now the question is what happens when you reverse the process – can you recover a higher quality signal form a lower quality file.
The answer I guess is – it depends.
If the algorithm is simple to reverse you should be able to re verse the process and recover the quality of the original (unless the algorithm is applied in a faulty way)
If the algorithm removes so much or reduces the quality of the original so much that it can’t be reversed or you need a very complex different algorithm to recover or restore the original you may be unable to improve the quality – a bit like photocopying a picture in black and white and then putting the black and white copy on the photocopier and hitting the colour copy button. You may be surprised to know this but there are algorithms that can try to do this but they are complex and fraught with problems. We all know that poor quality originals can be enhanced and brought closer to the original signal – for instance algorithms used by geographers and the military to improve satellite pictures. Or the “remastering” of old recordings.
So I think ( and I might be wrong) that if it is possible to convert something from a say 192kb/s bitrate back into wav format with say 1412kbit/s rate by reversing the process of the algorithm I have a higher quality signal (and my ears tell me this) You can, in fact, store the low bitrate format on the CD – it doesn’t automatically change back into a larger file when you burn it unless further processing is applied - check it out.

When it comes to sound ultimately we have to listen and judge it on that – and as said before it depends on the reproduction system that converts digital to the analogue sound we hear also.


Kahlau, I am interested in your response but being the intermediary is awkward :cool:
 
#12 · (Edited)
Well, Andante, that's quite some reply from your friend. I can find fault with almost nothing in it ... although I am curious (and mildly suspicious) about the suggestion of algorithms that can replace lost digital information. I don't know whether or not this is true. I just know that everything I've ever read about lossy formats being burned to CDs indicates that the digital bits of information that were discarded during the encoding process can't be recovered. Beyond that, it's beyond my understanding, too.

As an aside, ask your friend to download and set up foobar2000. It has a double-blind listening test .dll component as standard (called ABX Comparator). If your friend truly believes that he or she can accurately hear the differences between compressed and uncompressed sound files, this test will prove how good is his or her hearing. It'll also calculate the probability of guess work. Trust me, this test sorts the men from the boys. ;)

FK
 
G
#13 ·
I will pass on the message ;) , I have d/l 3 of the programs that you mentioned, foobar included but am still figuring out how to use them, one point that if I have understood correctly is: that when compressed to say mp3 certain digital info is removed and not just compressed so it could never be recovered, so why does the file size double ?? it must be adding something:cool:
 
#14 ·
#15 ·
Original digital recordings can be made in a number of different formats, all of which can be and are saved as wav files.
For instance I recently received the master wav files of an album originally recorded on 2" 24 track analog tape.
These had been recorded into Pro Tools, a high end digital audio workstation, and burned to DVD as 24 bit 96k audio wavs.
They could have been saved as 8, 16, 32, 44.1 (cd standard bitrate) 48,96, or 192 with obvious impact on the file sizes involved.
In order to get the best possible recording onto a cd it is (due to the Red Book standard as to how CD`s should be burned) only possible to use 16/44.1 rate.
Cd`s do not support anything higher, so that the resultant CD will play on a standard audio CD player.
So by definition CD quality is far less than is technically possible & indeed these days most recordings are made with higher sample and bit rates.
And this is where the tricky part comes in.
When the high quality recording is dropped or "dithered" down to standard CD format, there is inevitably some loss of signal information, but bear in mind this is information that your home system is physically incapable of reproducing even if it were on the CD in the first place.
So you are not actually losing big chunks of music unless you have a system capable of reproducing the higher rate files.
Funnily enough, a lot of what you perceive to be missing is our old friend dynamic range.

And in conclusion sample rate and bit rate are both part and parcel of the same situation.
A CD is by definition limited to the standard bit and sample rates, so the only variable is how efficiently the algorithm used to dither down the higher quality original recording file manages the job.

And no, as far as I am aware there are no reverse engineering programs that will allow the revesing of the lossy compression methods used to encode mp3, ogg vorbis etc.

It`s sort of like those bottomless Guinness bottles in the Irish jokes.
And if you ever find one, I`d like two as well.

(grin)

Ivan
 
#16 ·
Things are as Mr. Terrible says. Even though we've become accustomed to thinking of CDs as the best sound we can get, formats like SACD, DVD-Audio and others have shown what more there is to hear. And with some sites like Linn Records and Gimell Records now selling downloads in studio master quality, we're almost into an age where we can expect everything that's possible from digital sound reproduction.

FK
 
G
#17 ·
Where I have been going wrong is thinking that the employment of format converters I.e. Media Monkey Foobar etc to convert mp3 to FLAC and EAC would restore mp3 to an acceptable audio format whereas if I have understood you correctly it will make not a scrap of difference.
Now if I rip from a CD and convert to FLAC or EAC then burn, the result will be much better than a normal Window Media Player copy or WAV.
How’s that?

Is SACD popular I just do not come across it over hear:confused:
 
#18 ·
You're almost there, Andante.

Let's break this down into some simple (and rather too simplistic) equations:


WAV = Good as it gets / CD quality

FLAC = Same as WAV, but smaller filesizes

EAC = Not a format, but a program to accurately extract data from CDs

SACD = Super Audo CD / Better than CD quality


Does this help at all?


And yes, burning MP3s to CD does diddly squat to the sound quality. Once digital information is lost, it can't be recovered.

FK
 
G
#19 ·
OK, one other thing, mp3 is a small file due to both compression and removal of digi information, what is the digi info that is removed?? and would it be too inconvenient with to days tech advances to stop removing info :)
I have yet to hear a SACD they are dual layer I believe, are they better than Audio DVD?
I reckon its about time the d/l quality of Classical in particular was improved, ;)
 
#20 · (Edited)
MP3 files are compressed precisely because digital information has been removed - although this is only one form of compression, as I'll explain in a minute. The information removed relates to the frequencies which the human ear can't actually hear at either end of the sound spectrum; and the lower the bitrate used for encoding, the more data is removed from either end until the sound becomes very poor indeed.

When you use good-quality hifi equipment and compare the playback of an MP3 file to that of an SACD, you can hear clearly what's happened as a result of that compression. What you perceive when listening to SACD is much greater dynamic range, and consequently, a real sense of 'space' or presence in the recording. Some SACDs are so good that, depending on how advanced your equipment is, you can experience the music almost exactly the way the conductor would've heard it.

No compressed - or 'lossy' - format such as MP3 can ever give you what SACD can. This is the limitation of MP3: the technlogy is already very old (in techie terms), and can't be redesigned to remove less data. That's why we have WMA, OGG Vorbis, AAC, M4A, ATRAC3Plus and a myriad of other lossy formats, most of them better at compression of sound without such severve quality loss. But now that digital storage media is so cheap and plentiful, and broadband speeds are making larger files quicker to download, compression should slowly become a thing of the past.

Now, to complicate and confuse matters a little, there's another kind of compression. The FLAC encoding technology is very different to that of MP3, because digital data isn't 'thrown away' during compression. With FLAC, the compression works just like it does with a zip file. So, let's say that in our digitally ripped music track, we have the following binary sequence:

00000011110000111111

FLAC technology will look at this sequence and instead of seeing it in such a linear fashion, will effectively 'rewrite' the sequence thus:

6 x 0 4 x 1 4 x 0 6 x 1

Do you see the difference? The technology isn't trying to write great long strings of binary data, but instead 'compressing' these by writing them more intelligently and efficiently. This means that when FLAC files are decoded for playback, a 'reversal' of this intelligent compression results in sound quality IDENTICAL to that of the source CD; no information was discarded, remember, it was just written differently. ;)

As for DVD-A, the sound is comparable to that of SACD, although my preference is for the latter. I just think SACD has the edge, but I'm not sure there's a great deal in it.

I agree that classical downloads should be lossless (like FLAC), and I'm pleased to say that it's starting to happen. If you look at a site called Passionato, you'll see they sell classical music in FLAC format ... though only to UK customers at the moment, I believe. :( For historical recordings, Pristine Classical is another site offering FLAC (though one wonders why, given how poor some of the aged transfers were to begin with).

FK
 
#21 ·
Great post. It's not really that complicated, but few who understand it are willing to explain it in non-nerd language. The key thing is to transfer as much original material from the recording session to listeners' ears as possible. As Kuhlau says, with HD space so cheap and fast broadband connections becoming the norm in First World countries, there's increasingly little excuse to use the old lossy MP3 format.
 
G
#22 ·
As Kuhlau says, with HD space so cheap and fast broadband connections becoming the norm in First World countries, there's increasingly little excuse to use the old lossy MP3 format.
Hi 99, this is what I was trying to say in my non nerdy way:cool:

Kuhlau, thanks I am at last beginning to understand, it is something that puzzled me but being old and doddery I am easily confused:confused:
I suppose mp4 is not a great improvement, so all in all a case to be had for purchasing your music from your local dealer, at least for the present.
 
#26 ·
All computing still reduces to binary at its most basic level, Andante.

Interestingly, there was a response to an article in the letters pages of International Record Review earlier this year which basically gave the same technical 'advice' as I gave here ... yet I only read that back issue today. Nice to have my understanding independently confirmed. :)

FK
 
#27 ·
Confusing terms

I think part of the problem with this subject is the use of comfusing terms that have different meanings in different contexts -- the worst culprit being 'compression'

Compression is most often used to mean the control of dynamic range -- specifically the reduction of dynamic range, so that loud sounds aren't as loud as they were, while quieter sounds are louder. The idea being to let you hear the quite stuff above the background noise of the listening room while the loud bits don't annoy the neighbours.

However, in the context of digital audio codecs, 'compression' refers to the amount of data reduction being applied -- or how much the file size is reduced.

There are four basic ways to achieve a reduction in the data file size. The first is to use a lower sample rate. For example, instead of using the 44.1kHz rate of CD, the telephone system uses an 8kHz sample rate. The advantage is that you imediately have only about one fifth of the data to worry about... but the disadvantage is that the audio bandwidth is reduced to just 3.5kHz instead of 20kHz. Lousy for quality music, but adequate for intelligible speech... which is why it is used.

The second technique is to reduce the wordlength -- how many bits are used to describe the audio amplitude. Studio quality recordings are made with 24 bits per sample which gives a potential dynamic range in excess of 130dB. CDs are made with 16 bits per sample giving a potential dynamic range of over 90dB -- but that reduction in wordlength removes more than 30% of the data. Telephone systems use only 8 bits, reducing the data even more. But clearly, the disadvantage with this approach is a higher and more obvious noise floor, and a lower potential dynamic range.

The third option is to remove what is referred to as 'redundant data' and this is the scheme used by loss-less codecs like FLAC and MLP (amongst many others). If you consider the digital encoding of a photograph, each pixel must be described in terms of its colour and brightness, and in a high resolution picture that results in a huge amount of information. A loss-less codec reduces much of that information by avoiding repetition of identical data -- doing away with the 'redundant' information.

So if the sky is a uniform blue, say, rather than saying this is a bright blue pixel, this is a bright blue pixel, this is a bright blue pixel, this is a bright blue pixel... The codec notes it as this is a bright blue pixel, and so are the next 327, this is a less bright blue pixel and so are the next 28... and so on.

In this way, the amount of data required to describe the picture is reduced significantly, but the information and accuracy is preserved absolutely -- which is why it's called Loss-less. Audio signals tend to be very cyclical in nature much of the time, and this is largely predictable. The loss-less codecs simple take advantage of this 'predictability' to remove the redundant data and this reduce the file size. There are obvious limits to how far this can be taken, and that's why loss-loess audio codecs typically only reduce file sizes by about 60%.

The final option is where it all gets very contentious, and that's to remove 'irrelevant data' -- the elements of a complex sound recording that the human hearing system can not resolve and detect. Formats like Dolby Digital (AC3), DTS (apt-x), MiniDisc (ATRAC), DAB radio, MP3, MP4 and all the rest are lossy codecs and they all work (albeit in slightly different details) by trying to throw away parts of the sound that their designers' think we can't hear and won't notice.

The premise is that the human ear/brain is not a linear sound analysing system, but it actually a highly reactive and non-linear system whcih suffers from an effect called 'frequency masking'. Essentially, if I played you a quiet single-pitch tone at, say, 3kHz you would be able to perceive it quite happily. If I then add in a very much louder tone, starting at 100Hz, and gradually increase its frequency, you'll be able to hear and distinguish the two separate tones. But as the louder tone starts to approach about 2kHz the 3kHz tone will seem to disappear, and it won't come back until the loud tone has carried on up to something like 5kHz or more. It's quite a sobering demonstration, actually! But this 'frequency-masking' effect occurs throughout the hearing range and is the underlying principle of the old analogue tape noise-reduction systems like Dolby B and C.

In the case of the ever-popular MP3 format, the basic scheme is to divide the original audio signal into a large number of narrow frequency bands. The signal in each band is then analysed to determine which frequency components can and can't be heard in the presence of all the rest. Those that the systems thinks can't be heard are immediately discarded, never to be heard again! Obviously, this process immediately removes quite a lot of data from the signal.

The next stage is to analyse the signals elements that are left to determine how much louder they are than the system noise floor in each frequency band. The wordlength used to describe their amplitudes is then reduced to the minimum possible consistent with retaining a reasonable signal-noise ratio. This removes a whole load more data... and that's basically how the stereo signal from a CD player can be reduced from 1411kb/s (2x16x44100) down to a puny 128kb/s for MP3.

To trained ears, the quality loss is obvious, but the majority of the iPod listeners seem quite happy with the result, even though over 90% of the original information has been discarded.Clearly, using a higher data rate -- such as 320kb/s -- means that less data has to be thrown away and so the quality loss (and file size reduction) is not as great.

MP3 is a very clever system, but this is an area of technology that is advancing rapidly as the way the ear/brain works becomes better understood, and data processing power and speed improves. So MP3 is already old-hat, and there are better codecs around now (AAC being one of many) -- but the familiarity or MP3 is such that it will remain popular for a long time to come.

If you start with an MP3 file, a large proportion of data has been thrown away, as I've explained, and that can't be recovered in any way. However, if you convert the MP3 file to some other format (eg a wav file to burn onto an audio CD), the data has to be 'recompiled' into the appropriate format. Which means that instead of describing the data in terms of a series of narrow frequency ranges with variable wordlengths, the signal has to be rebuilt into conventional samples with a fixed wordlength... and in doing that considerably more data is required to describe the remaining audio.

But if you were to analyse that rebuilt signal carefully, you would see narrow frequency bands disappearing and reappearing as the signal changed, and the noise floor would be bouncing up and down in the different bands -- all as a result of the MP3 processing. So the quality remains at the MP3 level, even though the actual file size has increased.

Sorry for the long post -- but hopefully that's helped to explain what is going on in these systems and why they work in the way they do.

Hugh
 
G
#29 ·
Hugerr
Thank you for taking the time and effort to make such an informative post, I have a greater understanding now of the issues involved, it is to be hoped that this will be perused by our members.
Thanks again Kahlau for being so patient with me. :)
 
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