On 25/07/15 13:13, Jim web wrote:
In article <[email protected]>, Budgie
<[email protected]> wrote:
On 22/07/15 09:33, Jim web wrote:
In article <[email protected]>, Budgie
<[email protected]> wrote:
2) You may find that simply passing the file though ffmpeg or avconv
adds in sequential timestamps. i.e. using
ffmpeg -i infile.<ext> cleanfile.<ext>
To start at the beginning and why this is on GiP list is that all the
problem downloads are from GiP and are BBC 3 classical music downloads
such as The Early Music Show or Opera on 3 and they are mostly "hi fi"
stereo or supposed to be!
FWIW I use gip a lot to fetch R3 files and don't get such problems. I
usually play them with Audacious without any problems.
The problems I have date from the various times when my GiP setup was
flawed or atomic parsley not installed, all through my ignorance. They
fall into two categories, those that play but must be played from
beginning to end (no time markers,) and those which do not play at all
and stall the control app. (I am using BubbleDS control app and Linn
Majik renderer playing files served by Minimserver from NAS.)
So presumably your problem may be down to the player/renderer or something
odd about your setup when the files were fetched.
FWIW I keep files on a NAS and play them via access using a standard
network filing system. Just use general purpose programs - audacious for
audio and VLC for AV. Fairly basic and simple.
The files are delivered by my GiP setup as .m4a files and there is no
video but there is usually a thumbnail and useful metadata about the
particular recording which I would like to retain.
OK, I tend to use --no-tag which I think means to discard any tagging. But
I may be wrong about that.
The simple command with ffmpeg you gave above halved the size of the
file but it would not play. I tried various other variations, some
using winff GUI but none worked.
Ah. Be warned that using the command with no codec specifications at all
might change the codec/payload. You should use one of the ways we've
already described to specify keeing the codec payload unchanged. e.g. the
way I add
-acodec copy -vcodec copy
or the alternatives people have explained.
If you don't do that ffmpeg might have changed the content to some defaults
it thinks appropriate for you. Hence the change in file size.
Alternatively, the files may have junk in them for some reason I don't
know.
Do your playing programs report the codec, bitrate, etc, so you can see
what they think they are getting?
If you or others have time I would appreciate some further help and it
may be appreciated if I went off group or started a new thread until I
can announce success!
I can't say what others feel about this being relevant. But if it isn't
welcome here you could ask on a newsgroup like uk.rec.audio since your
interest is audio for this.
ffmpeg is accompanied by probe and play programs. If you have these one
will give info on a file. The other will try to play it. If it plays it
will confirm that ffmpeg can make sense of it even if your usual players
can't.
If all else fails for the broken files you could try using ffmpeg to
generate an LPCM wave file as its output. If that works it is easier to
check, and could then be converted to something like flac for use if the
problem is otherwise unsolvable.
Afraid the above is only general comments as I'm not clear on the cause of
your problem. I haven't used the other things you mention so have no idea
what they may have done.
Jim
Hi Jim,
Many thanks. I shall now do a check to see if these files play over the
lan on a laptop. My setup is not particularly exotic but I agree the
control app may be the issue in some cases.
I ran ffprobe on the first offending file and cannot see anything wrong.
Trouble is I have no idea what to look for. Will go back over above
advice in list. I am sure the data is in the file so in worst case I
could go down the LPCM route but not ready yet.
Will have a look at uk.rec.audio too.
Many thanks again.
Budge
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