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

Tue Jun 28 03:28:32 MSD 2011

Steven,

AFAIR, one of the recording app (not sure which) does not support mod_native_file, meaning for instance you can't dump your call using codec XXX to a raw file.
That's a quite old limitation, so perhaps it's not an issue anymore.

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Le 28/06/2011 à 00:10, Steven Ayre a écrit :

> Its not just voice quality. It's also CPU.
> 
> If you're using a compressed format, you're doing transcoding step. FS needs to decompress the file to L16 so it can compress it to the codec. Storing as wav containing l16 would avoid that. You can also use mod_native_file to compress the audio file to each codec format so that you don't need to transcode at all at the time because it's already done, although that does mean a file for each codec so more space needed.
> 
> Steve on iPhone
> 
> On 27 Jun 2011, at 20:43, Wes <wes-fs at 499x.com> wrote:
> 
>> 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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