[Freeswitch-users] .wav vs .gsm file sizes for recording calls.

[Wes]

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 at 499x.com 
> <mailto:wes-fs at 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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