Hello list,
I am about to have to dump Asterisk in favor of some other
VOIP/PBX solution; the reason? I have 304 voice prompts recorded as 22Khz
wav format files that sound like crumpling paper whenever I convert them to
the 8Khz wav/gsm format required by Asterisk. I was considering trying the
G.729 codec, but reading through the specs, I see that the 8Khz conversion
is going to dump me into the same pile of dung. Any body have any
suggestions?
Thanks
Danny Nicholas
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20101015/6a5d1a1a/attachment.htm
On 10/15/2010 09:59 AM, Danny Nicholas wrote:> > Hello list, > > I am about to have to dump Asterisk in favor of some > other VOIP/PBX solution; the reason? I have 304 voice prompts > recorded as 22Khz wav format files that sound like crumpling paper > whenever I convert them to the 8Khz wav/gsm format required by > Asterisk. I was considering trying the G.729 codec, but reading > through the specs, I see that the 8Khz conversion is going to dump me > into the same pile of dung. Any body have any suggestions? > > Thanks > > Danny Nicholas >hiring someone to re-record 304 prompts is not simpler and far faster than redeploying an entire system ? sounds like about a 4hr job. or find a better converter. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101015/19bee465/attachment.htm
pbx$ man sox
allpass frequency[k] width[h|k|o|q]
Apply a two-pole all-pass filter with central frequency
(in Hz) frequency, and filter-width width. An all-
pass filter changes the audio's frequency to phase
relationship without changing its frequency to amplitude
relationship. The filter is described in detail in [1].
~
Andrew "lathama" Latham
lathama at gmail.com
* Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software
* Learn more about Linux http://en.wikipedia.org/wiki/Linux
* Learn more about Tux http://en.wikipedia.org/wiki/Tux
_____
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of jon pounder
Sent: Friday, October 15, 2010 9:10 AM
To: asterisk-users at lists.digium.com
Subject: Re: [asterisk-users] drop dead fix
On 10/15/2010 09:59 AM, Danny Nicholas wrote:
Hello list,
I am about to have to dump Asterisk in favor of some other
VOIP/PBX solution; the reason? I have 304 voice prompts recorded as 22Khz
wav format files that sound like crumpling paper whenever I convert them to
the 8Khz wav/gsm format required by Asterisk. I was considering trying the
G.729 codec, but reading through the specs, I see that the 8Khz conversion
is going to dump me into the same pile of dung. Any body have any
suggestions?
Thanks
Danny Nicholas
hiring someone to re-record 304 prompts is not simpler and far faster than
redeploying an entire system ?
sounds like about a 4hr job.
or find a better converter.
Option 2 is what I have in mind (BTW, with the "talent" I have, your 4
hrs
is closer to 80, after normalizing, trimming and "prodding").
What I do now is record the file using soundrec, normalize it with
Audiograbber, then trim it with Audacity before converting it with Sox.
Which of these is letting me down, (or it is "the loose nut on the
keyboard")?
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20101015/55ba4d87/attachment.htm
On Fri, 15 Oct 2010, Danny Nicholas wrote:> Hello list, > > I am about to have to dump Asterisk in favor of some other > VOIP/PBX solution; the reason? I have 304 voice prompts recorded as 22Khz > wav format files that sound like crumpling paper whenever I convert them to > the 8Khz wav/gsm format required by Asterisk. I was considering trying the > G.729 codec, but reading through the specs, I see that the 8Khz conversion > is going to dump me into the same pile of dung. Any body have any > suggestions?Why are you converting them to GSM? Why not convert them to the technology you're using for your phones and trunks? That would be much more efficient. (If you're using g729 for trunks, then that will sound better as GSM to g729 conversion does sound bad) Or maybe it's your conversion software? What are you using? Gordon
On Fri, 15 Oct 2010, Danny Nicholas wrote:> ????????????? I am about to have to dump Asterisk in favor of some other > VOIP/PBX solution; ?the reason?? I have 304 voice prompts recorded as > 22Khz wav format files that sound like crumpling paper whenever I > convert them to the 8Khz wav/gsm format required by Asterisk.? I was > considering trying the G.729 codec, but reading through the specs, I see > that the 8Khz conversion is going to dump me into the same pile of > dung.? Any body have any suggestions?Can you post a link to a sample "before" and "after" file as well as the command line you are using to convert the file? -- Thanks in advance, ------------------------------------------------------------------------- Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000
I never had this problem, and this is certainly not asterisk's fault. Probably your conversion is not good. Can you email me a file and I'll do conversion on my end, and if sounds good, let you know how I did it. Then a script can be written to convert them all. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-10-15 10:25 AM, "Steve Edwards" <asterisk.org at sedwards.com> wrote: On Fri, 15 Oct 2010, Danny Nicholas wrote:> I am about to have to dump Asterisk in f...Can you post a link to a sample "before" and "after" file as well as the command line you are using to convert the file? -- Thanks in advance, ------------------------------------------------------------------------- Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101015/ee0a9d67/attachment.htm
-----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Steve Edwards Sent: Friday, October 15, 2010 9:21 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] drop dead fix On Fri, 15 Oct 2010, Danny Nicholas wrote:> ????????????? I am about to have to dump Asterisk in favor of some other > VOIP/PBX solution; ?the reason?? I have 304 voice prompts recorded as > 22Khz wav format files that sound like crumpling paper whenever I > convert them to the 8Khz wav/gsm format required by Asterisk.? I was > considering trying the G.729 codec, but reading through the specs, I see > that the 8Khz conversion is going to dump me into the same pile of > dung.? Any body have any suggestions?Can you post a link to a sample "before" and "after" file as well as the command line you are using to convert the file? The sox line I am using (version 14.0.1) is Sox foo.wav -r 8000 -c 1 bar.wav resample -ql Before I found Audiograbber I used this line Sox -v 2 foo.wav -r 8000 -c 1 bar.wav resample -ql
On 10/15/2010 08:59 AM, Danny Nicholas wrote:> Hello list, > > I am about to have to dump Asterisk in favor of some other > VOIP/PBX solution; the reason? I have 304 voice prompts recorded as > 22Khz wav format files that sound like crumpling paper whenever I > convert them to the 8Khz wav/gsm format required by Asterisk. I was > considering trying the G.729 codec, but reading through the specs, I see > that the 8Khz conversion is going to dump me into the same pile of > dung. Any body have any suggestions?In addition to all the other comments you've received (including the fact that Asterisk does not "require" GSM format files), keep in mind that *any* product that plays these files over the PSTN is going to have to downsample them to 8KHz and, at a minimum, use G.711 companding. That is what the PSTN uses, so it's not possible to have higher fidelity than that. There were some comments in other replies about your files being 'quiet' (low average volume level)... this won't help your situation at all, because it means that any artifacts caused by resampling and compression/decompression will end up at a relatively high amplitude compared to the original signal (resulting in a low signal-to-noise ratio), and when the listener increases the volume level on their listening device, the noise level will be increased along with it. For these sorts of tasks, you really do want the source material recorded at a fairly high volume level. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kfleming at digium.com Check us out at www.digium.com & www.asterisk.org