Andrew Howe suggested:
> A bit of a cop-out, but try using ffmpeg?
Same as avconv - "Unknown" quality and immediate crash of mp3 player.
Re Chris Liddell's point, the sample rate is the standard 44100: these
are ordinary CD tracks. Naxos specialise in cheap CDs, I would not
expect them to do anything fancy involving high res audio.
Leo said:
> The output of:
> avconv -i original.flac
> and
> avconv -i converted.mp3
> will give you detailed info on the files, e.g. what sample rate is used.
Original flac:
Duration: 00:02:50.13, bitrate: 613 kb/s
Stream #0.0: Audio: flac, 44100 Hz, 2 channels, s16
mp3 from avconv:
Duration: 00:02:50.16, start: 0.000000, bitrate: 145 kb/s
Stream #0.0: Audio: mp3, 44100 Hz, stereo, s16, 32 kb/s
mp3 from ffmpeg:
Duration: 00:02:50.16, start: 0.000000, bitrate: 145 kb/s
Stream #0.0: Audio: mp3, 44100 Hz, stereo, s16, 32 kb/s
ogg from audacity:
Duration: 00:02:50.13, start: 0.000000, bitrate: 117 kb/s
Stream #0.0: Audio: vorbis, 44100 Hz, stereo, s16, 128 kb/s
I get the impression that the options I was using apply only to video
files, and that ffmpeg and avconv just do not have equivalent options
for audio.
Experiment shows that the player is happy with an unconverted flac
file and with an ogg produced using audacity, which does what I tell
it to do - quality 4 gives 128kb/s. In principle then, I can do what I
want with audacity. This might not be convenient without a CLI - but,
I find that it has a batch processing system called Chains. I will see
if I can make that work. As far as I can tell from the audacity help
system, it should do.
Thanks to all for the comments. I still think something I could use
from the command line would be easier - a simple shell script loop.
Peter Alefounder.
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