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Franklin  
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 More options Jul 7 2007, 6:00 pm
Newsgroups: alt.comp.freeware
From: Franklin <franks...@nomail.com>
Date: Sat, 07 Jul 2007 23:00:22 +0100
Local: Sat, Jul 7 2007 6:00 pm
Subject: Re: flac to mp3
On 07 Jul 2007, hummingbird <hummingb...@2die4.com> wrote:

> On Thu, 05 Jul 2007 18:54:04 +0100  'Franklin' posted this onto
> alt.comp.freeware:

>>On 05 Jul 2007, dadiOH <dad...@guesswhere.com> wrote:

> [snip]

>>> The whole point of my post was to point out that the conversion
>>> of one format - lossy or not - requires that the format being
>>> converted be decoded to wave before/during conversion.

>>> The person to whom I was replying seemed to feel that decoding
>>> something to wave and then re-encoding it resulted in a worse
>>> re-encoding than "doing it in one step".  He said...

>>> "By "higher level of fidelity", I am saying that a higher level
>>> of quality will be obtained in this example from a single
>>> transcoding operation than from two transcoding operations."

>>> and that simply is not correct since he was talking about going
>>> to wave as the first operation,  to lossy format as the second.

>>Dadioh, what you say is in principle perfectly correct but in
>>practise it does not always seem to work out so straightforwardly.

>>I've just posted to someone in ACF about Audacity and referene a
>>thread which discusses how using Audacity on an incoming WAV and
>>then saving the WAV can result in a material change to the WAV on
>>account of "side-issues" when the sample rate gets changed.

>>See http://tinyurl.com/2t6p39  Odd huh?

> Nothing odd about it Franco, it's probably a straightforward
> malfunction within Audacity when it changes the sample rate of a
> wav file. I don't have that problem when I use Cool Edit - but
> that's a payware program.

> From the debate you linked to (dated Jan/2006):

> "The key appears to be the way that Audacity converts from one
> sample rate to another."

> I'd agree with that comment at first pass.

> But none of that changes what dadiOH says that there is no such
> thing as a one-step conversion from Flac to MP3 ...you have to go
> thru wav en route even though it might appear transparent to the
> end user.

There is a difference between:
(a) converting from flac to WAV and then taking the WAV and converting
it to MP3
(b) converting from flac to MP3.

(a1) The sampling rate and bit depth of the WAV may be mismatched with
respect to the sampling rate and the bit depth specified for the flac.  
And similarly, the sampling rate and bit depth of the WAV may be
mismatched with respect to the sampling rate and the bit depth specified
for the MP3.

http://extra.benchmarkmedia.com/wiki/index.php/Sample-rate shows  
distortion (from quantisation errors) caused by sample rate conversion.

(a2) When one says WAV then one usually means PCM which is, of course,
an uncompressed format.  However WAV can very often contain ADPCM rather
than PCM and ADPCM is a compressed format, albeit only very slightly
compressed.  Assumptions about the PCM encoding scheme inside the
created WAV may not be true because many people consider ADPCM WAVs to
be good enough to be a top class WAV.  For ex. MP3 players which record
often create ACPCM encoded audio which they glibly call WAV.  That idea
wouldn't work here.

(b1)  Of course (b) uses an intermediate internal representation of the
audio but this representation would be consistent with the input and
output in a way that (a1) or (a2) is not.

As for the OP's question in this thread, it seems the extra time is a
glitch in Audacity.  I too don't get that result in my $$$ware audio
editors.


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