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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