Bit rate from --aactomp3

Jim web web at
Sun May 31 09:16:28 PDT 2015

In article <97444A64ED014661A5C9CBA90C579E3F at vasonote>, Vangelis
<northmedia1 at> wrote:
> On Sun May 31 11:22:16 BST 2015, RS wrote:

> > Is there any way... to get get_iplayer to respect the --radiomode
> > flashaaclow option when --aactomp3 is used?

> Hello Richard... Short answer: NO You may refer to this past post of
> mine:
> The main change since then is that with the "Audio Factory" overhaul,
> the "flashaaclow" (as well as the "hlsaaclow") radiomode is now encoded
> in HE-AACv1 (not v2, so no Parametric Stereo) and the SR is 48kHz
> (previously at 44.1kHz).

> On Sun May 31 12:27:41 BST 2015, RS wrote:

> > Once it has been compressed to 48kbit/s, any higher frequency
> > information is gone. It cannot be retrieved by converting at a higher
> > bit rate.

> We do have our own audio expert in this list (Jim web), who graces us
> with most excellent posts, and I stand to be corrected, but what you
> state is true for the first encode from the original uncompressed file.

FWIW a low bitrate doesn't *have* to mean that "any higher frequency
information is gone". It all depends on the 'settings' for the judgements
the lossy encoder was given to apply when thinning out the data. What will
have happened is a loss of detail. That said, the BBC do in my experience
tend to preferentially discard 'high frequency' details as that is felt to
have less impact than losing the same amount of data from low frequencies.
In effect, they target the bits allocated to the frequency range where our
ears notice problems more easily.

What is certainly true is that you can't get back what was discarded.

As a general rule, I'd avoid transcoding one lossy format into another. If
you have to, I'd go for a higher rate for the 'destination' to try and
avoid more loss as far as possible. Particularly if you're going from aac
to mp3. However this isn't really a matter of "improving the result" but
"minimising further harm". :-)

I can't comment on if 128 kbps mp3 is 'good enough' for avoiding more
harm to 48 kbps aac as I've never tried it. Not sure I'd want to. 8-]

In practice when I have had to transcode aac or mp3 I convert it to LPCM
(wave) or flac so I'm losing as little as possible. In effect the result
sounds like the source. Has its flaws, but without additional damage.

I'm not sure from the earlier postings, but I guess the problem is that mp3
is needed by the end-device (player) because it can't render aac. If so,
I'd go for a high rate for the mp3 even though it may seem daft. But then
personally I'd probably bin the mp3 device and get one that can play aac.

I don't know about the podcasts, though, as I haven't tried them. I just
get the programs I want as broadcast.

Sorry if I've misunderstood the situation.


Armstrong Audio
Audio Misc

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