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
> 
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Hi Vangelis,
Very many thanks for your instructive and helpful reply.  I shall
research the issues you highlight.  Following dinkypumpkin's advice I
ran ffmpeg on the problem file but have not yet been able to try the new
file on the Linn player.

Thanks for the info on MediaInfo etc.  This will help me investigate the
several other files which are causing me problems.

Finally, for the spoken word I would be happy to use mp3.  This is how I
receive BBC podcasts of spoken word such as In Our Time and these are
fine.  If I wish to transcode the problem HE-AACv2 file to mp3 should I
do this with ffmpeg or another program?

Many thanks again,
Budgie