You're right I think. :) Was talking about playback... guess it was pretty
late.
Post by David PonzoneSteven,
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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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
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>
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.
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>
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
<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
Post by WesIn 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?
RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz
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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