On Thu, 2 Apr 2015, Pavel Troller wrote:

 I'm disappointed with functionality of the "file convert" command. Its
results are very poor, at least in ast11. It looks that the conversion
is performed through something like slin8 or even alaw/ulaw. There is
a huge difference between listening to for example MP3Player() using
a G.722 codec and Playback() of the G.722 file prepared by "file convert"
from the same MP3 file.
 I was looking at res_convert.c. It's unbelievably simple :-). The trick
must be somewhere in the underlying functions. I tried to analyze them as
well but I never noticed anything like specification of the format used for
the conversion. It just reads frames from one file and writes them to the
other one. But which frames ? Who specifies the format ? I'm confused.
 I would like to implement a simple logic to choose proper "intermediate
format" according to the combination of input/output file types. A small
hint to push me forward would be greatly appreciated :-).

Just my 'personal biases...'

I always thought having Asterisk convert files was dumb. I will never use it.

I'd much prefer a 'stand-alone' command line utility like sox.

Why would you want to use a hammer instead of a scalpel?

Do you really want to drag all the functionality and bloat of a general purpose audio conversion tool into your PBX?

Do you really want to explain to your boss why you crashed the PBX during an important shareholder conference because you wanted to convert the new music on hold and you hit a bug in all the new bloat?

'Do one thing, do it well, and move on.'

Are sox plugins for telephony codecs viable?

I'll stop ranting now. You caught me on a frustrating day.

--
Thanks in advance,
-------------------------------------------------------------------------
Steve Edwards       sedwa...@sedwards.com      Voice: +1-760-468-3867 PST
Newline                                              Fax: +1-760-731-3000

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-dev mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-dev

Reply via email to