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

Wes wes-fs at 499x.com
Tue Jun 28 00:45:44 MSD 2011


I'm sorry, but what is the SLIN format?  I tried that extension and it's 
invalid.  I searched the wiki and found nothing.

On 6/27/2011 3:29 PM, David Ponzone wrote:
> 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.
> Any efficient compression takes CPU, and you may want to avoid doing 
> that live on your server.
>
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> Le 27/06/2011 à 21:43, Wes a écrit :
>
>> 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 <http://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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