<html><head></head><body bgcolor="#FFFFFF"><div>Its not just voice quality. It's also CPU.</div><div><br></div><div>If you're using a compressed format, you're doing transcoding step. FS needs to decompress the file to L16 so it can compress it to the codec. Storing as wav containing l16 would avoid that. You can also use mod_native_file to compress the audio file to each codec format so that you don't need to transcode at all at the time because it's already done, although that does mean a file for each codec so more space needed.</div><div><br>Steve on iPhone</div><div><br>On 27 Jun 2011, at 20:43, Wes <<a href="mailto:wes-fs@499x.com">wes-fs@499x.com</a>> wrote:<br><br></div><div></div><blockquote type="cite"><div>
<meta content="text/html; charset=ISO-8859-1" http-equiv="Content-Type">
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 class="moz-txt-link-freetext" href="http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session"><a href="http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session">http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session</a></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"><a href="http://wiki.freeswitch.org/wiki/Weekly_Conference_Call#Past_Calls">http://wiki.freeswitch.org/wiki/Weekly_Conference_Call#Past_Calls</a></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 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"><a href="mailto:wes-fs@499x.com">wes-fs@499x.com</a></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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