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Which ATRAC bit rate corresponds to a specific Mp3 bit rate?

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Nicolas1400

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Everybody knows the best way to store atrac files on our discs is

PCM/APE/FLAC/AAL--->ATRAC

But sometimes, we have a great difficult-to-find album in mp3 format, and we cannot get the original (loseless) format of it.

I´d like to make a conversion bitrate table, where we can say what´s the best bitrate option (atrac) for every mp3 bitrate we EVENTUALLY could have. (Yes, i really HATE mp3)

The idea is to store the mp3 audio in atrac files, getting the best balance quality/disk space.

For example, it would be very stupid to store mp3@128 as atrac@256 (we´ll get extra disk space without any quality improvement).

As a start... let say, mp3@128 --> atrac@64 (as i read in the forums)

mp3 bitrate - best atrac bitrate option

128 64

160 .

192 .

224 .

256 .

320 .

VBR?? ???

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Transcoding from any lossy format to another causes further loss - call it generation loss.

Transcoding from any bitrate [lossy or not] to a lower one causes more significant and noticeable loss.

The plain truth is that taking something that already is artifacted by one compression algorithm and transcoding it to another simply takes the original signal, artifacts and all, and adds more artifacting to it - regardless of the originating and destination compression formats.

The only way to avoid really noticeable generation loss is to transcode to the most transparent [i.e. highest encoding quality] bitrate available.

Thereby, the best bitrate option would be the highest available, HiSP or a3+ 352kbps [depending on whether you're using SS 3.3 or not], unless you want to ensure that there's -no- further loss, in which case use PCM.

As much as you may hate MP3, if quality/size is important to you, then keep the original MP3s. Transcoding to a higher bitrate is wasting space and still adding further loss. Transcoding to a lower bitrate is saving space, and adding much further loss.

So, to answer directly -

MP3 - a3+

--------------

128 - 256/352

160 - 256/352

192 - 256/352

224 - 256/352

256 - 256/352

320 - 256/352

VBR - 256/352

As I said, if you prefer not in incur any further loss, simply replace all of the above with "PCM". If you're running SS 3.3, replace all of the above with a3+ 352kbps.

Keep in mind, though - what one person finds acceptable in terms of artifacting/loss is bound to be different from another person's opinion. My opinion is based on the assumption that you don't want things to be any worse than they were when they started. The best way to avoid this is, of course, to not transcode the original files in any way.

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I'd agree with dex Otaku. I reckon you should keep the original somewhere, even stored on a CD somewhere. Then keep a ATRAC copy in your library.

If your library process/system, changes in the future it would be good to have the original encoding somewhere. If you plan on keeping a Hard Disk Library, I'd always recommend buying two large disks. One as a backup, archive and one as the live library. 200/250GB disks are cheap these days, and personally I find the convience of extra space worth it. I also archive some stuff on DVDRW or CDRW too.

I have a mix of mp3's and WAV's in my library. I tend to change my mind on what I want every few months and usually up the bitrate as I get better audio equipment. Then I'll recode some of my original stuff at the higher bitrate. For example, initially I was happy with my analog recordings from my cassettes, but now I plan to look for CD's of fav albums 2nd hand. Just to get the better quality.

As my HiMD isn't my primary player, I tend to encode from CD rather then from my MP3 library when I want to put something on a MD. Lately this was to experiement with 352 and it meant ripping to WAV/PCM then encoding to 352.

I think your library and bitrate should be set up to best suit the player you use most.

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Interesting topic...

my recommendation in general is to buy you fav songs original Audio CDs 2nd hand from ebay, and then save them as:

==> WAV (PCM, not compressed, RAW Audio)

or

==> FLAC (compressed, lossless)

or

==> OGG (Quality level 10, 250 kbits/s per Channel)

I also hate MP3 due to the variety of quality options... there can be different 160 kbits/s per channel in MP3 if you have studied the the lame encoder...

but most of the people dont know that, and just simply use MP3 programs in which its been told "128 kbits (CD Quality)"...

so just never download any MP3 from internet, make your own MP3 from original RAW audio, for example using these parameters in lame:

for STEREO

-b 320 -m s -o -q 0 -k

for MONO

-b 160 -m m -o -q 0 -k

thats the only way to get the best quality from MP3s...

anyone disagree ?

Edited by DJ_THE_CROW
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What you said is true, you always will loss some information when you transform from a compress format to another compress one.

But for a simple answer for the initial question, of makinkg a bit rate list to compare both formats, will be:

MP3 -- A3,A3+

128 -- 64 (in some cases, sometimes MP 128 sounds better)

160 -- 132

192 -- 132

224 -- 256

256 -- 256

320 -- 256

VBR -- 256

and I making this comparation just hearing both, not using any software test or wave analisys, etc.

I compare MP3 with A3 and A3+ at 64, 132, 256 Bit rate. the ones I can use in Hi-MD, of course if I I say that 256 is comparable, 352 will be better.

I know many people will have a different opinion, but this is what I hear.

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DJ The Crow,

I would disagree with your MP3 encode settings, but only slightly

LAME uses lossless joint stereo (its not the Psycho acoustic version) therefore I would always suggest that Joint Stereo is used as it gives better compression for the same quality OR better quality for the same compression, as in this case.

Your stereo line would change to

-b 320 -m j -o -q 0 -k

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Qwakrz: While I agree with you in principle, there are exceptions to this.

M/S stereo [the type of joint stereo encoding used by most lossy codecs] is optimal for use with 2-channel sources that have a high degree of coherence between channels - i.e. much is the same in each channel, such as most vocal tracks in music recordings and other things you could consider to be monaural [hence the M in M/S - mid/side] in the mix. One encoded channel contains coherent signals [M] while the other is incoherent signals [s, out of phase signals]. [Yes, I'm oversimplifying this, but you'll see the general point in a moment]

The big advantages to this method are - 1, If a mono signal is recorded, 100% of the alotted bitrate can go to just the one signal, rather than half to each stereo channel in the case of discrete stereo encoding, and 2, phase coherence between channels should be more easily maintained.

There are times when this is NOT what you want, though - such as if you are recording 2 channels that have little or no coherence between them. An example of this would be recording from two separate lapel mics during an interview; the level of coherence [at 0 degree phase diff as well as 180 degree phase diff] should be pretty close to zero at all times. Discrete stereo coding with this signal would be more efficient and likely less artifacted.

Just my extra $0.02.

Cheers.

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I've read several things on this board about converting from MP3 to ATRAC. I've also looked over a thread in the "News" section that (d)evolved into a debate between lossy and lossless, and frankly, that thread hurts my brain.

Simple question:

If you convert a 128/160/192 lossy MP3 into a "lossless" WAV file, is there more audio data magically created in the WAV file? (or, does the WAV file sound better than the original MP3?)

Several threads have suggested converting MP3's to an Audio CD (or a virtual Audio CD) and then using Simple Burner to copy it to the minidisc player. I'm just wondering if this is because the final ATRAC file ends up sounding better...?

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GIGO - Garbage In, Garbage Out.

Reiterated: you can't improve on what's not there.

When you convert MP3 to any lossless format, all yuo're doing is basically testing the quality of the encoding, for one, and of the decoder you're using, for another. Whatever artifacts were in the file to begin with will be in the lossless file afterwards.

There are times when transcoding can be done more easily from a CD image than from individual files.

One example of this is if you have a CD image in MP3 format with a cuesheet; the cuesheet will allow you to make a CD with the same loayout as the original, and the fact that the MP3 is a disc image [a single contiguous file that has all the audio from the beginning to the end of the originating CD with no gaps] allows you to make a gapless CD [or CD image to mount and rip with SS] despite MP3's general non-conformity to the same framelength as CD.

While I've never used MP3 files this way, I've used disc images sourcing from FLAC and WavPack files to make gapless HiMD copies. Basically, with any format that your CD writing software can understand but SS can't [which covers a lot of ground], whether a disc image or not, sometimes it's just the most convenient way to get things to a HiMD or MD - rather than converting those files to WAV, importing into SS, transcoding, &c. In many cases it's really just a matter of preference which way you go; the end results will be the same.

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