Radio File Format Questions

[Vangelis forthnet]

On Sat Jul 13 15:52:02 BST 2013, Budgie wrote:

>As usual, a couple of questions.
>
>Is the file format HE-AAC v2 the normal output for a low bit rate
>download or is it another, to me, anomaly?

 Hello.
 Yes, 'flashaaclow' radiomode yields an audio file packaged in an MP4 
container
(whose format profile is "Apple audio with iTunes info", hence the .m4a 
extention),
which in it contains a raw ADTS (audio data transport stream) .aac file 
encoded in
HE-AACv2 as you correctly state; HE-AAC is AAC+SBR, v2 indicates the use of
PS (parametric stereo). The encode uses a VBR with a mean value of 48kbps.
NB that if you come from a non-UK IP, this is the only audio quality 
available to you
for National Stations.
 If in the UK, the default high quality mode (= flashaac/flashaacstd) is 
again an
.m4a file, but the audio stream contained therein is encoded in AAC LC (no 
SBR, no PS)
@ 128kbps (320kbps for Radio 3) ABR, that's why it is more compatible with 
software/
hardware players.
 Depending on the player used, the PS part may be skipped (audio plays in 
mono), or both
PS+SBR skipped, in which case audio plays in mono and in very low quality, 
since only
half the sampling rate is used.
 In my Windows setup I haven't come across a software player that does not 
play at least
the AAC part of a HE-AACv2 encode. But hardware players (like your network 
player here:

http://www.linn.co.uk/all-products/network-music-players/sneaky-ds

) behave differently; the features list of yours only mentions a "generic 
AAC" decoding support,
so it may be expected that it does not support HE-AAC (try a World Service 
download) or
HE-AACv2, as you have found out.

On your laptop, any ffmpeg based software player (FFplay, + the ones you 
mentioned)
can play fully HE-AACv2 audio streams.

>What programme can I use to find out the detailed information of what is
>in each .m4a file?

As a generic multimedia file "investigator", you can use the CLI FFprobe,

http://ffmpeg.org/ffprobe.html

which, together with FFplay, is part of the FFmpeg package - if it isn't 
available
for your OS, maybe its fork "avprobe" is:

http://libav.org/avprobe.html

As a personal choice though, I'd recommend MediaInfo - it comes both as a 
GUI & CLI
and is available for a plethora of OSes, including yours (openSUSE 12.2) 
here:

http://mediaarea.net/el/MediaInfo/Download/openSUSE

>what would you recommend I run to change
>the format of the sound file and to what format?

dinkypumpkin in your answer to you has kindly suggested a recode from
HE-AACv2 -> AAC-LC through FFmpeg (or avconv). If your FFmpeg
is built with support for one of the non-free AAC encoders (libfaac or the
far better libfdk_aac), then I guess it'd be fine,
but the native encoder (-c:a aac -strict -2)
lacks in performance, especially in music parts -
for speech is fine.

If I can humbly share my opinion, I have found that a transcode from
HE-AACv2 @48kbps -> LAME MP3 @ 96kbps (for spoken content) /
112 (or even 128) kbps (for music content) is more than adequate and I would
propose that, since your SneakyDS does play MP3 files.

Regards.

Vangelis