Radio File Format Questions

[Budgie]

On 14/07/13 13:37, Vangelis forthnet wrote:
> 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

Hi Vangelis,
I have been working on other stuff and only now return to sort out my 
problem files.  Would you have time to help some more please?

First question concerns diagnosis of the files which do not play on Linn 
device.  I do have ffprobe but do not see any mention of HE-AACv2 or 
AAC-LC when I run it on a problem file.  What I do see is :-

[CODE]
Stream #0:0(und): Audio: aac (mp4a / 0x6134706D), 44100 Hz, stereo, 
fltp, 48 kb/s (default).
[CODE]

I interpret this to mean that I have the downloaded the flashaaclow 
version which from your advice I interpret to be the HE-AACv2 encoded 
version.  Is that correct?

Since I have no idea why I get these files from time to time I assume I 
do not have get_iplayer set up correctly so that is cannot download in 
this format.  First request then is what should I put in my options file 
to ensure I only get the higher quality radio options?

Second question is please could you help with ffmpeg command line to 
convert these files to files that will play as suggested previously by 
dinkypumpkin.  I regret my knowledge is not up to doing it without more 
help with command options.

Finally I note I could transcode on downloading and save these files as 
mp3 files but is it also possible to transcode to flac?

Grateful for any further help when you have time.
Regards,
Budgie