VoIP Mailing List Archives

[andy at fabulous4.co.uk]

Hi, Sorry but I just can't find this in the documentation. I'm using recordFile to record incoming messages. I'd like the audio files produced to be 11025Hz rather than 8kHz is this possible? What setting do I need to change? Many thanks Andy

[brian at freeswitch.org]

No its not possible yet. Voicemail has an option to record in 11025 but not arbitrary rates for recording otherwise.

/b

On Jun 12, 2009, at 9:37 AM, Andy wrote:

[anthony.minessale at g...]

Looked easy enough so i added
record_sample_rate variable that should influence it for you if you set it in advance.
 


On Fri, Jun 12, 2009 at 9:49 AM, Brian West <brian@freeswitch.org (brian@freeswitch.org)> wrote:

[andy at fabulous4.co.uk]

Excellent, thanks Anthony, I'll give it a go. From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Anthony MinessaleSent: 12 June 2009 17:03To: freeswitch-users@lists.freeswitch.orgSubject: Re: [Freeswitch-users] Sample rate and recordFile Looked easy enough so i added record_sample_rate variable that should influence it for you if you set it in advance. On Fri, Jun 12, 2009 at 9:49 AM, Brian West <brian@freeswitch.org (brian@freeswitch.org)> wrote: No its not possible yet. Voicemail has an option to record in 11025 but not arbitrary rates for recording otherwise. /b On Jun 12, 2009, at 9:37 AM, Andy wrote: Hi, Sorry but I just can't find this in the documentation. I'm using recordFile to record incoming messages. I'd like the audio files produced to be 11025Hz rather than 8kHz is this possible? What setting do I need to change? Many thanks Andy________ Brian West brian@freeswitch.org (brian@freeswitch.org) -- Meet us at ClueCon! Freeswitch-users@lists.freeswitch.org (Freeswitch-users@lists.freeswitch.org)MSN:anthony_minessale@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email])GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])IRC: sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])pstn:213-799-1400

[msc at freeswitch.org]

On Fri, Jun 12, 2009 at 9:26 AM, Andy <andy@fabulous4.co.uk (andy@fabulous4.co.uk)> wrote:

[andy at fabulous4.co.uk]

Hi All, I've tested this new variable and everything works grand. I've tested recording to wav,mp3 and shoutcast and in all cases the sample rate is set correctly. I was about to post an entry on the wiki but I discovered a very similar variable already there called record_rate. I've tested this and it doesn't appear to work. Would you like me to replace this entry with details of the new one that does seem to work? A few more questions: 1) I notice that when I change the sample rate it automatically changes the bit rate too. I understand why this is the case but wondered if it was just as easy to be able to control the bitrate as well as the sample rate. 2) When I use a sample rate other than 8000 I get a warning 'Sample rate doesn't match'. I guess this puts some extra load on the server. If all my calls are being recorded and all at 11025 can/should I alter the sample rate of the base call to 11025? Many thanks for sorting this one for me and for all your help. regards Andy From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Michael CollinsSent: 12 June 2009 18:52To: freeswitch-users@lists.freeswitch.orgSubject: Re: [Freeswitch-users] Sample rate and recordFile On Fri, Jun 12, 2009 at 9:26 AM, Andy <andy@fabulous4.co.uk (andy@fabulous4.co.uk)> wrote: Excellent, thanks Anthony, I'll give it a go. Andy, can you report back on your success with this variable? Also, we would appreciate it if you could add an entry to the wiki on the channel_variables page. Let me know if you have any questions and I'll be glad to help.-MC

[brian at freeswitch.org]

On Jun 18, 2009, at 11:54 AM, Andy wrote:

[mrene_lists at avgs.ca]

Most calls are at 8kHz. The formula for bandwidth is sampling rate * bit rate * channels, so 8000 * 8 * 1 = 64000 = 64kbps (per way).

Math

On 18-Jun-09, at 1:01 PM, Andy wrote:

[brian at freeswitch.org]

The call rates we support are 8, 16,32 and 48k

/b

On Jun 18, 2009, at 1:01 PM, Andy wrote:

[brian at freeswitch.org]

look in mod_shout you'll see my calculations.. I think it has to be multiples of 16 if I recall.

/b

On Jun 18, 2009, at 1:01 PM, Andy wrote:

[andy at fabulous4.co.uk]

Thanks Brian, So, just to calrify will the base call always be 8kHz? On a related note, do you happen to know the bitrate of each open channel/live call? Is it 16 kilobits per second like the recorded audio? I need to do some calculations on the badwidth required to handle a certain number of concurrent calls. Many thanks Andy From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Brian WestSent: 18 June 2009 18:11To: freeswitch-users@lists.freeswitch.orgSubject: Re: [Freeswitch-users] Sample rate and recordFile On Jun 18, 2009, at 11:54 AM, Andy wrote: 1) I notice that when I change the sample rate it automatically changes the bit rate too. I understand why this is the case but wondered if it was just as easy to be able to control the bitrate as well as the sample rate. If you're talking about mod_shout, NO. You'll end up picking an invalid bitrate and asking why it doesn't work... been there done that... I changed it a few months back to pick the optimal bitrate for the sample rate. 2) When I use a sample rate other than 8000 I get a warning 'Sample rate doesn't match'. I guess this puts some extra load on the server. If all my calls are being recorded and all at 11025 can/should I alter the sample rate of the base call to 11025? NO. Your phone call is running at 8kHz, Your sound file is 11025 and they don't match, If you were to play this file into an 8k channel without a resample it would sound a little like satan. or a dragging tape deck. The file has to be resampled to match the current session rate. /b

[dave at 3c.co.uk]

- plus UDP/RTP overhead. Budget 10 calls/megabit for G.711 and you'll
have a bit of headroom available.

--Dave

[anthony.minessale at g...]

or go over the limit and you'll have Max Headroom =D


On Thu, Jun 18, 2009 at 1:16 PM, David Knell <dave@3c.co.uk (dave@3c.co.uk)> wrote: