Requests For Features
web at audiomisc.co.uk
Sun Jul 26 01:11:50 PDT 2015
In article <55B3C44D.7010702 at errichel.co.uk>, Budgie
<ajebay at errichel.co.uk> wrote:
Comments prompted by the Linn reply...
> Hi Chris, I gave up on this subject due to pressure of work back at the
> end of 2013 and had forgotten how far I had reached. It appears that
> for the GiP Radio 3 downloads I messed up there are no time markers and
> it is the Linn Player which needs them. Linn advised me as follows:-
> AAC files store data about the media stream they contain within their
> outer MPEG4 container. Seekable AAC files will also provide a collection
> of offsets allowing an AAC decoder to jump to a variety of points in
> the media stream, enabling seeking.
It is possible, I guess, that the version of ffmpeg you're using does have
a flaw. You could try taking an existing LPCM wave file you have and using
ffmpeg to create an m4a from it. Then seeing if the Linn player will play
it or not if you treat it as if you'd got it via gip. For comparability,
choose a 48k/stereo/16bit file to make your test m4a.
If that *doesn't* work, then the implication is that either something is
strange about your copy of ffmpeg or the Linn player isn't very clever. To
clarify that, see if it will play with alternative software using your
computer. FWIW I use Audacious, but I'm sure there are many alternatives.
You could also try a more up-to-date version of ffmpeg. (Assuming you're
not already doing so.)
> The file you provided us with appears to contain only 1 offset: the
> offset of the start of the audio track within the file (after all the
> metadata). Therefore, there are no further offsets that can allow our
> AAC decoder to safely move to another point within the stream.
> As a solution to this, we would recommend using a ripper (e.g.
> dbpoweramp) to transcode the track to another format for use with the DS.
> I concluded and hoped from this there was no need to transcode to
> another format if I can just "transcode" to the same format. I just
> need to get the ffmpeg command correct.
Yes, I think you are correct. If you wish to avoid any loss of detail I'd
recommend sticking to either 'recontainering' (using the 'copy' options of
ffmpeg as described previously) or converting to a 'wave' file (i.e. LPCM)
or a flac file.
> Will keep trying. I am not sure if the -idx options in mplayer will do
> the job but can give it a try.
I don't have much experience with mplayer but I'd hope it should play the
files OK. Be interested to see your results.
Armstrong Audio http://www.audiomisc.co.uk/Armstrong/armstrong.html
Audio Misc http://www.audiomisc.co.uk/index.html
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