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

Tue Jun 28 00:29:54 MSD 2011

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.

David Ponzone  Direction Technique
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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 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> 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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