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.
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session
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)
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.
On 6/24/2011 6:31 PM, Michael Collins wrote:
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.
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. (http://wiki.freeswitch.org/wiki/Weekly_Conference_Call#Past_Calls)
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:
<2831>:ls
-1s conf_call_2011-06-15.*
18736
conf_call_2011-06-15.mp3
23044
conf_call_2011-06-15.ogg
199756
conf_call_2011-06-15.wav
<2832>:file
conf_call_2011-06-15.mp3
conf_call_2011-06-15.mp3:
MPEG ADTS, layer III, v2, 24 kBits, 16 kHz, Monaural
<2833>:file
conf_call_2011-06-15.ogg
conf_call_2011-06-15.ogg:
Ogg data, Vorbis audio, mono, 16000 Hz, ~48000 bps, created by:
Xiph.Org libVorbis I
<2834>:file
conf_call_2011-06-15.wav
conf_call_2011-06-15.wav:
RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit,
mono 16000 Hz
Note that the file sizes are in 1K blocks.
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.
-MC
On Fri, Jun 24, 2011 at 2:26 PM, Wes
<wes-fs@499x.com>
wrote:
In my tests, if I record a call in .wav format, a 10 second
file is
about 177,000 bytes, while a 10 second .gsm file is 17,000
bytes.
I then used sox to convert the .gsm file to a .wav file, and
it stayed
at around 17,000 bytes. So, is the default recording format
for .wav
using a higher sample rate? vs the default conversion format
for the sox
tool?
checking the file type using "file" I see that the larger one
is:
RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit,
mono 8000 Hz
and the wav created by sox via the default conversion from
.gsm is:
RIFF (little-endian) data, WAVE audio, GSM 6.10, mono 8000 Hz
So apparently the larger wav file is 16 bit... how are these
recording
parameters controlled? Can I set it to record directly into
the smaller
wav format? Or will I have to run sox on every file...
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