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I'm sorry, but what is the SLIN format? I tried that extension and
it's invalid. I searched the wiki and found nothing. <br>
<br>
On 6/27/2011 3:29 PM, David Ponzone wrote:
<blockquote cite="mid:D279722B-D927-4F78-8557-3A8CEAAE5516@ipeva.fr"
type="cite">You should really consider recording in SLIN, and
then, batch compressing that later (during off-peak hours) to MP3
if you really need it to be smaller.
<div>Any efficient compression takes CPU, and you may want to
avoid doing that live on your server.</div>
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<div>Le 27/06/2011 à 21:43, Wes a écrit :</div>
<br class="Apple-interchange-newline">
<blockquote type="cite">
<div bgcolor="#ffffff" text="#000000"> I guess part of my
confusion here was due to the term "raw data" mentioned
in conjunction with the .gsm extension on the wiki page
below... but actually gsm is a compressed format.<br>
<br>
<a moz-do-not-send="true" class="moz-txt-link-freetext"
href="http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session">http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session</a><br>
<br>
So, what is the best "compressed" format to use for
recording voice (that is available as a direct recording
format inside freeswitch)? There are tons of formats
listed when I do "show file", but I tried a few and they
are also giving me large files like the wav extension
did. (au, for example)<br>
<br>
Even though the PCM/Wave format is preferred for voice
quality, when we're talking about a 10:1 compression
ratio, if the sound quality is still acceptable, I'd
rather just record directly into the compressed format.
We're talking about ~10- 20 minute recordings that will
need to be transferred over the internet to a third
party.<br>
<br>
On 6/24/2011 6:31 PM, Michael Collins wrote:
<blockquote
cite="mid:BANLkTimxdvjSU=gDXMLYLb17jDwamWDSbw@mail.gmail.com"
type="cite">I would caution you to consider adding
disk space before you try to compress all your
recordings. The 16 bit SLIN that FS normally puts in
your wave files are pretty easy to handle, whether
playing back in a FS session, or encoding for playback
on some other device. <br>
<br>
An alternative might be to use lame to convert them to
MP3's or ogg/vorbis files. If you look on the main FS
conf call page you'll see I have the weekly recordings
in multiple formats. (<a moz-do-not-send="true"
href="http://wiki.freeswitch.org/wiki/Weekly_Conference_Call#Past_Calls">http://wiki.freeswitch.org/wiki/Weekly_Conference_Call#Past_Calls</a>)<br>
<br>
Here are some stats for last Wednesday's call. Note
that I record wave files in 48kHz then use sox to
downsample to 16kHz wave, then I convert that 16kHz
file into MP3 and Vorbis (in an ogg container). Here's
what the results look like:<br>
<br>
<span style="font-family: courier new,monospace;"><2831>:ls
-1s conf_call_2011-06-15.*</span><br
style="font-family: courier new,monospace;">
<span style="font-family: courier new,monospace;"> 18736
conf_call_2011-06-15.mp3</span><br
style="font-family: courier new,monospace;">
<span style="font-family: courier new,monospace;"> 23044
conf_call_2011-06-15.ogg</span><br
style="font-family: courier new,monospace;">
<span style="font-family: courier new,monospace;">199756
conf_call_2011-06-15.wav</span><br
style="font-family: courier new,monospace;">
<br style="font-family: courier new,monospace;">
<span style="font-family: courier new,monospace;"></span><span
style="font-family: courier new,monospace;"><2832>:file
conf_call_2011-06-15.mp3 </span><br
style="font-family: courier new,monospace;">
<span style="font-family: courier new,monospace;">conf_call_2011-06-15.mp3:
MPEG ADTS, layer III, v2, 24 kBits, 16 kHz,
Monaural</span><br style="font-family: courier
new,monospace;">
<br style="font-family: courier new,monospace;">
<span style="font-family: courier new,monospace;"></span><span
style="font-family: courier new,monospace;"><2833>:file
conf_call_2011-06-15.ogg </span><br
style="font-family: courier new,monospace;">
<span style="font-family: courier new,monospace;">conf_call_2011-06-15.ogg:
Ogg data, Vorbis audio, mono, 16000 Hz, ~48000 bps,
created by: <a moz-do-not-send="true"
href="http://Xiph.Org">Xiph.Org</a> libVorbis I</span><br
style="font-family: courier new,monospace;">
<br style="font-family: courier new,monospace;">
<span style="font-family: courier new,monospace;"></span><span
style="font-family: courier new,monospace;"><2834>:file
conf_call_2011-06-15.wav </span><br
style="font-family: courier new,monospace;">
<span style="font-family: courier new,monospace;">conf_call_2011-06-15.wav:
RIFF (little-endian) data, WAVE audio, Microsoft
PCM, 16 bit, mono 16000 Hz</span><br
style="font-family: courier new,monospace;">
<br style="font-family: courier new,monospace;">
Note that the file sizes are in 1K blocks.<br>
<br>
So, bottom line is this: if you have the disk space
then use wave. If you don't have disk space for wave
then get some! :D If you REALLY need to use a
different format then choose something like MP3 or
Vorbis for long-term storage. <br>
<br>
-MC<br>
<br>
<div class="gmail_quote">On Fri, Jun 24, 2011 at 2:26
PM, Wes <span dir="ltr"><<a
moz-do-not-send="true"
href="mailto:wes-fs@499x.com">wes-fs@499x.com</a>></span>
wrote:<br>
<blockquote class="gmail_quote" style="margin: 0pt
0pt 0pt 0.8ex; border-left: 1px solid rgb(204,
204, 204); padding-left: 1ex;"> In my tests, if I
record a call in .wav format, a 10 second file is<br>
about 177,000 bytes, while a 10 second .gsm file
is 17,000 bytes.<br>
<br>
I then used sox to convert the .gsm file to a
.wav file, and it stayed<br>
at around 17,000 bytes. So, is the default
recording format for .wav<br>
using a higher sample rate? vs the default
conversion format for the sox<br>
tool?<br>
<br>
checking the file type using "file" I see that the
larger one is:<br>
RIFF (little-endian) data, WAVE audio, Microsoft
PCM, 16 bit, mono 8000 Hz<br>
<br>
and the wav created by sox via the default
conversion from .gsm is:<br>
RIFF (little-endian) data, WAVE audio, GSM 6.10,
mono 8000 Hz<br>
<br>
So apparently the larger wav file is 16 bit... how
are these recording<br>
parameters controlled? Can I set it to record
directly into the smaller<br>
wav format? Or will I have to run sox on every
file...<br>
<br>
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