Radio File Format Questions

[Vangelis forthnet]

On Sun Jul 14 22:48:46 BST 2013, Budgie wrote:

>Very many thanks for your instructive and helpful reply.

 You are most welcome; apologies I did not get back to you sooner.
I am currently mourning the sudden loss of an external USB HDD
1TB in capacity, which was full with close to 900GB worth of files
(the majority of which are irreplaceable), accumulated in the course of
nearly 3 years. It will cost me a small fortune to buy a replacement disk
and recover about 85% of the data that can be salvaged from the
failed disk :-( :-(

>This will help me investigate the
>several other files which are causing me problems.

 More often than not, the first "Listen Again" radiomode that is made
available online by the beeb after a radio show has finished is the
"flashaaclow" one (in GiP's terms), which they call the "Lower
Bandwidth Version". I don't know if this is by design, but it's my
personal observation. For example, after the Official Chart Show
on Radio 1 finishes at 19:00 BST on a Sunday, the "flashaaclow" mode
appears first 1,5-2 hours later, while it usually takes another 1-2 hours 
for
the "flashaacstd" mode to be made available (this depends on the load on
the encoders chain...).
 If you are doing your audio downloads in an automated way via a PVR
list and you have not explicitly asked for the "flashaacstd" mode, then, 
when
your PVR list is executed, it may sometimes download the low quality
version of a programme, because at that time it was the only one available.
This is possibly why you end up with some HE-AACv2 .m4a files inside
your downloads folder...

>for the spoken word I would be happy to use mp3.  This is how I
>receive BBC podcasts of spoken word

FYI, these are mono files encoded @ 64kbps constant bitrate (CR); the
source used for the encode is probably a high quality master, that's why
they do sound quite good, but TBH I do not like the monaural sound,
especially when listened to through headphones.

>If I wish to transcode the problem HE-AACv2 file to mp3 should I
>do this with ffmpeg or another program?

Any audio conversion software capable of fully decoding HE-AACv2 and
encoding to mp3 should do the task; I have no clue what are your choices
in your platform (OpenSuSE 12.2).
 FFmpeg is fine for this - I would use something like this:

ffmpeg -i foo.m4a -vn -c:a libmp3lame -ab 96k -ac 2 -ar 44.1k foo.mp3

(NB that the m4a file's metadata will be lost during the conversion)

On Mon Jul 15 18:31:47 BST 2013, Budgie wrote:

>Yes I am using ffmpeg.  First try gave me twice the file
>size as advised by Vangelis but I am looking at options to set lower bit
>rate for output file.

 You never mentioned that file size is an issue for you.
HE-AACv2 is a very efficient encoder at low bitrates (22-64kbps), that is 
why
it is now used very widely for internet radio streaming (to cut down on 
bandwidth
costs). In comparison, LAME MP3 lacks considerably in this field.
If you are prepared to compromise with some quality loss, you could 
experiment
with bitrates lower than 96kbps, or try a variable bitrate (VBR) scheme, but 
I
wouldn't try values lower than 64kbps (unless you are about to listen to the 
end
result on a mobile phone with a cheap set of earphones...)

Just my two "eurocents" ...

Vangelis.