16-bit PCM audio is just "raw" audio, sometimes you'll see "SLIN" or "signed linear" or some other designations. Whenever you see that just know that FS is dealing with raw, unencoded audio. Therefore it is neither alaw nor ulaw. You *can* record to a file with specific encoding, but that's usually not needed. 99% of the time just a simple wav file with PCM audio is perfectly sufficient.<br>
<br>-MC<br><br><div class="gmail_quote">On Thu, May 31, 2012 at 11:00 AM, Hector Geraldino <span dir="ltr"><<a href="mailto:Hector.Geraldino@ipsoft.com" target="_blank">Hector.Geraldino@ipsoft.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Thanks Peter, it's much more clear now.<br>
<br>
By default, when recording a session to a wav file, I can see in the audio file properties that: the sample rate is 8000 Hz (8kz as mentioned), and the codec says "Uncompressed 16-bit PCM audio". By default, is it ulaw or alaw?<br>
<br>
Sorry for the confusion, I'm trying to catch up with all this information about codec negotiation, codecs and audio formats but it's a little bit overwhelming for a novice like me.<br>
<br>
Thanks again<br>
<br>
-----Original Message-----<br>
From: <a href="mailto:freeswitch-users-bounces@lists.freeswitch.org">freeswitch-users-bounces@lists.freeswitch.org</a> [mailto:<a href="mailto:freeswitch-users-bounces@lists.freeswitch.org">freeswitch-users-bounces@lists.freeswitch.org</a>] On Behalf Of Peter Olsson<br>
Sent: Thursday, May 31, 2012 2:10 AM<br>
To: FreeSWITCH Users Help<br>
Subject: Re: [Freeswitch-users] uuid_record and recording output format<br>
<br>
The sample rate is the rate, so 8000 is the same as 8khz, it's got nothing to do with 8 or 16 bit.<br>
<br>
FreeSWITCH always converts to L16 (linear 16-bit), so that's why you get this result. You can force to record to a raw file, for instance, try recording to testfile.PCMA to record in G7.11 alaw format.<br>
<br>
/Peter<br>
<br>
30 maj 2012 kl. 16:37 skrev "Hector Geraldino" <<a href="mailto:Hector.Geraldino@ipsoft.com">Hector.Geraldino@ipsoft.com</a><mailto:<a href="mailto:Hector.Geraldino@ipsoft.com">Hector.Geraldino@ipsoft.com</a>>>:<br>
<br>
Greetings,<br>
<br>
I'm using a 3rd party application (ndev dragonmobile) to get the transcription of some audio recorded by FreeSWITCH. Think about it as a voicemail transcription service. The problem I'm facing is that, when I record a session using uuid_record, the output file is encoded in PCM 16-bit @ 8khz. Correct me if I'm wrong, but my understanding is that if I want to capture audio from calls coming from the PSTN (analog/landlines), the best I can do is to record it in 8-bits (using G.711).<br>
<br>
I don't want to use sox (or any other tool) to resample the output file, and what I've tried so far is setting the sample_rate variable on the diaplan as recommended on the wiki: <a href="http://wiki.freeswitch.org/wiki/Variable_record_rate" target="_blank">http://wiki.freeswitch.org/wiki/Variable_record_rate</a><br>
<br>
<action application="set" data="record_sample_rate=8000"/><br>
<br>
This doesn't have any effect on the generated wav file, which is still encoded in 16-bits. So my question is: does this variable affects the behavior of the uuid_record command? Or, do I really need to encode the audio output in 8-bits when the origin of the call comes from the PSTN? How is FreeSWITCH encoding the audio in 16-bits if, in theory, the best rate we can get from an analog line is 8-bits?<br>
<br>
Sorry if I'm misunderstanding something, but I'm not a telephony/voip guy, more like a java developer.<br>
<br>
Thanks for your help.<br>
<br>
<br>
!DSPAM:4fc62d7332761385138176!<br></blockquote></div><br>